Session Initiation Protocol Workshop

MITEL Session Initiation Protocol Workshop 3300 ICP Product Release 9.0 Issue 1 26 July 2009

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Session Initiation Protocol (SIP) Workshop Table of Contents SIP Overview PASTE TABLE OF CONTENTS FROM FIRST MODULE HERE Mitel SIP PASTE TABLE OF CONTENTS FROM SECOND MODULE HERE SIP Phones PASTE TABLE OF CONTENTS FROM THIRD MODULE HERE SIP Trunks PASTE TABLE OF CONTENTS FROM FOURTH MODULE HERE Maintaining and Troubleshooting PASTE TABLE OF CONTENTS FROM FOURTH MODULE HERE Appendix PASTE TABLE OF CONTENTS FROM FOURTH MODULE HERE i

Session Initiation Protocol (SIP) Workshop Student Code of Conduct Mitel University makes every effort possible to provide a safe, clean and professional environment for students attending training classes. It has become necessary, for the benefit of all students, to define what is expected from those attending classes at Mitel University. Please observe the following guidelines. Punctuality Unless otherwise specified, all classes begin at 9:00 a.m. Students are required to return from breaks and lunch promptly, as the instructor specifies. Instructors will begin lectures promptly at the scheduled times. Appropriate Behavior Students are expected to participate in class as professionals. Disruptive behavior will not be tolerated. Disruptive behavior is any action interfering with the instructor s presentation or action distracting from another student s ability to participate in the class. If, at the instructor s discretion, a student is being disruptive, the following steps will be taken: 1st Occurrence: Verbal warning. The student will be advised that his or her behavior is disruptive. 2nd Occurrence: Verbal warning. The student and the student s manager or supervisor will be informed that this is the final warning. 3rd and Final Occurrence: The student will be dismissed from the remainder of class and the student s manager or supervisor will be informed that the student has been released from class. The only option available to the student is to take the course exam at a proctored testing center, at the student s expense, or retake the course, in its entirety, at full tuition. No refund will be issued. Training Equipment Mitel University has made every effort to provide a state-of-the-art training facility and training equipment. Every effort has been made to provide the technology and equipment necessary to provide students with a real-world environment. All training systems and equipment (including PCs and the PC Network) are provided as tools to enhance the training experience. Equipment is only to be accessed and utilized for the completion of class lab exercises as the instructor indicates. Unauthorized exploring of, or experimenting with the training equipment will be considered disruptive behavior and will not be tolerated. Leaving Class Prior to the Final Certification Exam Occasionally, a student may have a bona fide reason to leave class early due to a family emergency, death in the family, etc. If a student must leave class prior to the administration of a final written exam for any reason, the only option available to the student is to complete the final written exam at a proctored testing center. Testing fees from the testing center are the responsibility of the student and/or company requesting the test. ii

Session Initiation Protocol (SIP) Workshop Course Description This practical workshop provides the student with hands-on experience of the Session Initiation Protocol (SIP) and its uses within the telecommunications environment. The workshop focuses on how Mitel implements SIP within an array of end user devices, intersystem trunks, external applications and Internet Telephony Service Provider trunks. It provides details of monitoring and troubleshooting devices and trunks by focusing on network packet captures. Objectives At the completion of the course you will be able to: Implement and configure SIP based phones on a Mitel 3300 ICP Implement and configure SIP trunks between Mitel 3300 ICPs and to a variety of 3rd Party providers. Monitor and troubleshoot a SIP registration and trunk communication Audience The audience for this course is Mitel 3300 ICP engineers who have an understanding of the 3300 ICP platform, a working knowledge of Voice over IP communications and who want to improve their understanding and practical knowledge of the Session Initiation Protocol. iii

Session Initiation Protocol (SIP) Workshop Resources During this course, you will be referring to the following resources: Course Manual Additional Software Virtual Machines iv

SIP Overview 1 Objectives When you finish this module, you will be able to: Describe the components associated with SIP Describe the workings of SIP

Mitel SIP Workshop 1-2 Module 1 - SIP Overview

SIP Overview Introduction SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions. It is an Internet standard published originally in March1999 as RFC 2543, but later superseded by RFC 3261 in June 2002. The primary function of SIP is to initiate and control sessions; communication channels between two or more devices. SIP can invite participants to already existing sessions and can also add or remove media from sessions. SIP is independent of the underlying transport protocols. SIP does not provide any services. It provides the framework for other protocols to use in order to establish a complete architecture. These protocols may include the Real-time Transport Protocol (RTP) (RFC 1889) for transporting real-time data and providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC 2326) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) (RFC 3015) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) (RFC 2327) for describing multimedia sessions. However the basic functionality and operation of SIP does not depend on any of these protocols. SIP is based on a request and response transaction model similar to HTTP. Each transaction consists of a request that invokes a particular method or a function on the server and at least one response. SIP provides the capabilities to: Determine the location of the target end point SIP supports address resolution, name mapping, and call redirection. Determine the media capabilities of the target end point Via Session Description Protocol (SDP), SIP determines the lowest level of common services between the end points. Conferences are established using only the media capabilities that can be supported by all end points. Determine the availability of the target end point If a call cannot be completed because the target end point is unavailable, SIP determines whether the called party is already on the phone or did not answer in the allotted number of rings. It then returns a message indicating why the target end point was unavailable. Establish a session between the originating and target end point If the call can be completed, SIP establishes a session between the end points. SIP also supports mid-call changes, such as the addition of another end point to the conference or the changing of a media characteristic or codec. Handle the transfer and termination of calls SIP supports the transfer of calls from one end point to another. During a call transfer, SIP simply establishes a session between the transferee and a new end point (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties. 1-3

Mitel SIP Workshop SIP Entities A SIP network is composed of four types of logical SIP entities. Each entity has specific functions and participates in SIP communication as a client (initiates requests), as a server (responds to requests), or as both. One physical device can have the functionality of more than one logical SIP entity. For example, a network server working as a Proxy server can also function as a Registrar at the same time. Following are the four types of logical SIP entities: USER AGENT In SIP, a User Agent (UA) is the endpoint entity. User Agents initiate and terminate sessions by exchanging requests and responses. RFC 3261 defines the User Agent as a logical entity that can act as both a user agent client and user agent server, as follows: User Agent Client (UAC) a client application that initiates SIP requests. User Agent Server (UAS) a server application that contacts the user when a SIP request is received and that returns a response on behalf of the user. Some of the devices that can have a UA function in a SIP network are: workstations, IPphones, telephony gateways, call agents, automated answering services. PROXY SERVER A Proxy Server is an intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced either internally or by passing them on, possibly after translation, to other servers. A Proxy interprets, and, if necessary, rewrites a request message before forwarding it. REDIRECT SERVER A Redirect Server is a server that accepts a SIP request, maps the SIP address of the called party into zero (if there is no known address) or more new addresses and returns them to the client. Unlike Proxy servers, Redirect Servers do not pass the request on to other servers. REGISTRAR A Registrar is a server that accepts REGISTER requests for the purpose of updating a location database with the contact information of the user specified in the request. 1-4 Module 1 - SIP Overview

SIP Overview Messages There are two types of SIP messages: Requests sent from the client to the server. Responses sent from the server to the client. REQUESTS A request invokes a particular method, or function, on the server and at least one response. Request Methods Method INVITE ACK BYE CANCEL OPTIONS REGISTER INFO Description Initiates a call, changes call parameters (re-invite) Confirms the final response for INVITE Terminates a calls Cancels searches and ringing Queries the capabilities of the other side Registers with the Location Service Sends mid-session information that does not modify the session state RESPONSES Response messages contain numeric response codes. The SIP response code set is partly based on HTTP response codes. There are two types of responses and six classes: RESPONSE TYPES Provisional (1xx class) provisional responses are used by the server to indicate progress, but they do not terminate SIP transactions Final (2xx, 3xx, 4xx, 5xx, 6xx classes) final responses terminate SIP transactions. CLASSES 1xx = provisional, searching, ringing, queuing etc. 2xx = success 3xx = redirection, forwarding 4xx = request failure (client mistakes) 5xx = server failures 6xx = global failure (busy, refusal, not available anywhere) 1-5

Mitel SIP Workshop Method Example Response Codes Description 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress 200 OK 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 410 Gone 413 Request Entity Too Large 415 Unsupported Media Type 416 Unsupported URI Scheme 408 Request Timeout 414 Request-URI Too Long 420 Bad Extension 421 Extension Required 423 Interval Too Brief 480 Temporarily Unavailable 481 Call/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 491 Request Pending 493 Undecipherable 500 Server Internal Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported 513 Message Too Large 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable MESSAGE PARTS SIP messages are composed of the following three parts: START LINE Every SIP message begins with a Start Line. The Start Line conveys the message type (method type in requests, and response code in responses) and the protocol version. The Start Line may be either a Request-line (requests) or a Status-line (responses), as follows: The Request-line includes a Request URI, which indicates the user or service to which this request is being addressed. This address can be re-written by proxies. The Status-line holds the numeric Status-code and its associated textual phrase. 1-6 Module 1 - SIP Overview

SIP Overview HEADERS SIP header fields are used to convey message attributes and to modify message meaning. They are similar in syntax and semantics to HTTP header fields (in fact some headers are borrowed from HTTP) and thus always take the format: <name>:<value> Headers can span multiple lines. Some SIP headers such as Via, Contact, Route and Request- Route can appear multiple times in a message or, alternatively, can take multiple commaseparated values in a single header occurrence. BODY (CONTENT) A message Body is used to describe the session to be initiated (for example, in a multimedia session this may include audio and video codec types, sampling rates etc.), or alternatively it may be used to contain opaque textual or binary data of any type which relates in some way to the session. Message bodies can appear both in request and in response messages. SIP makes a clear distinction between signaling information, conveyed in the SIP Start Line and headers, and the session description information, which is outside the scope of SIP. Possible body types include: SDP see Session Description Protocol (SDP). Multipurpose Internet Mail Extensions (MIME). Others to be defined in the IETF and in specific implementations. 1-7

Mitel SIP Workshop SIP Transactions A SIP Transaction consists of a number of messages that are sent and responded to in order to establish a session. A typical transaction will proceed as follows: User Agent Client UAC INVITE ------------------------> 100 Trying <----------------------- 180 Ringing <----------------------- 200 OK <----------------------- ACK -----------------------> Communication -------------------- BYE -----------------------> 200 OK <----------------------- User Agent Server UAS Session Establishment 1. The calling User Agent Client (UAC) sends an INVITE message to the User Agent Server (UAS). This message contains a Request URI and an SDP packet describing the media capabilities of the calling terminal. 2. The UAS receives the request and immediately responds with a 100 Trying response message. 3. The UAS starts ringing to inform the user of the new call. Simultaneously a 180 Ringing message is sent to the UAC. 4. The call is answered and the UAS sends a 200 OK message to the UAC. This message also contains an SDP packet describing the media capabilities of UAS device. 5. The UAC sends an ACK request to confirm that the 200 OK response was received. The session is established and other protocols, as described in the SDP Packet, are used to send the communication to each party. Session Termination 1. The caller decides to end the call and hangs-up. This results in a BYE request being sent to the UAS. The Message contains the Request URI used in the INVITE message. 1-8 Module 1 - SIP Overview

SIP Overview 2. The UAS responds with 200 OK message and notifies the user that the conversation has ended. As more devices are added to the network, the transactions become more involved and details can be changed at various levels. The next example shows how a proxy server can be used between the two user agents: User Agent Client UAC INVITE ----------------> 100 Trying <---------------- 180 Ringing <---------------- 200 OK <---------------- Communication <---------------------- Proxy Server INVITE ---------------> Trying <--------------- Ringing <--------------- 200 OK <--------------- Communication -------------------------> User Agent Server UAS BYE ----------------> 200 OK <---------------- Proxy Server BYE -----------------> 200 OK <----------------- This call flow shows how a Proxy Server can pass requests from one user agent to another. 1-9

Mitel SIP Workshop Session Description Protocol (SDP) SDP is the protocol used to describe multimedia session announcement, multimedia session invitation and other forms of multimedia session initiation. A multimedia session is defined, for these purposes, as a set of media streams that exist for a duration of time. SDP PACKETS SDP packets usually include the following information: Session information Session name and purpose. Time(s) the session is active. Since the resources necessary for participating in a session may be limited, it would be useful to include the following additional information: Information about the bandwidth to be used by the session. Contact information for the person responsible for the session. Media information Type of media, such as video and audio. Transport protocol, such as RTP/UDP/IP and H.320 Media format, such as H.261 video and MPEG video. Multicast address and Transport Port for media (IP multicast session). Remote address for media and Transport port for contact address (IP unicast session). 1-10 Module 1 - SIP Overview

SIP Overview Lab 1 Viewing SIP Packets in Wireshark In this lab you will: View a Wireshark capture of a SIP Session Note To complete this lab, you will need to have Wireshark installed. For details on Wireshark, please see Appendix B View a Wireshark Capture Step Task Expected Result/Observation 1 Browse to and open C:\Mitel Academy\Mitel SIP Workshop\SIP Trace 1.pcap 2 Select the textbox next to Filter: Type sip and press Enter This will filter all other packets leaving only SIP packets in the view. 3 Select Statistics > VoIP Calls 4 Select the call from the list and click Graph The graph shows a graphical representation of the SIP Transaction 5 Select each SIP message in turn (not the RTP packets) and view the details in the Packet Details section of the Wireshark tool 6 Close the Graph and VoIP Calls windows Remove the filter and look at the RTP Packets How many packets are there per second? 7 Close SIP Trace 1 and open SIP Trace 2 Repeat the exercise, this time try to figure out what is happening with the two calls 1-11

Mitel SIP 2 Objectives When you finish this module, you will be able to: Describe the workings of the Mitel SIP Compliance Program and the SIP Centre of Excellence Download the latest Interop Reference Guide

Mitel SIP Workshop 2-2 Module 2 Mitel SIP

Mitel SIP SIP Compliance Program In order to achieve best results, Mitel has created a SIP Compliance Program where service Providers, endpoints, gateways, firewalls, applications servers and enterprise SIP Servers can be tested according to defined standards to ensure compatibility. There are four levels: Mitel Approved Reserved for MSA Gold Preferred members only, this rare classification is reserved for key strategic components of our portfolio for which Mitel assumes the full responsibility for support, acting as the interface between the customer and the 3rd party as necessary. Compatible The most common certification which means the device/service has been tested and/or validated by the Mitel SIP CoE team. Product support will provide all necessary support related to the interop, but issues unique or specific to the 3rd party will be referred to the 3rd party as appropriate. 3 rd Party Self-Assessed Denotes that the 3rd party provider of the device/service has performed the interop testing they deem necessary and has asserted to Mitel that the interop was successful. All support related to the interop is referred directly to the 3rd party, as they are the ones making the assertion of compatibility, not Mitel. Field-Assessed For informational purposes only, field-assessed means that the device/service has been tested and/or used to some degree by someone successfully, though details may or may not be available. Mitel product support does NOT apply to field-assessed interops. Note An up-to-date list can be obtained from Mitel OnLine. 2-3

Mitel SIP Workshop SIP Centre of Excellence The SIP Centre of Excellence is a team of people dedicated to ensuring that Mitel continues to lead in SIP implementation, by innovating and testing new ideas, features and programs. In order to accomplish the goal of interoperability, Mitel have devised strict testing scenarios and guidelines that should be followed. These guidelines are available on Mitel OnLine and are composed of two sections namely SIP Phones and SIP Trunking. The following diagram shows the test plan that is used for SIP Trunks The tests cover a wide range of scenarios and include, among other tests: Basic Outgoing and Incoming Outgoing and Incoming Failures ONS, LS Trunks and PRI Interoperability Registration and Authentication Provisional Responses Privacy, Codecs, Hold, DTMF, FAX Transfers and Conferencing Long Ringing and Long Calls 2-4 Module 2 Mitel SIP

Mitel SIP Once test cases are performed and traces obtained, the details should be sent to sipinterop@mitel.com for approval. 2-5

Mitel SIP Workshop SIP endpoint (lineside) resiliency At present, there are no clear SIP standards for SIP resiliency that would be comparable to what is achieved in a proprietary implementation for devices connected to a 3300 ICP solution using MiNET such as the Mitel 5304 / 5312 / 5324 / 5330 and 5340 IP Phones. As such, Mitel s implementation features different levels of SIP resilient behaviour, ranging from basic resiliency in accordance with current specifications to resiliency comparable to what is achieved under MiNET. The level of resiliency supported is determined primarily by the capabilities of the endpoint. The four levels of resiliency available are: Bronze, Silver, Gold and Platinum. The lower levels are possible with generic SIP endpoints through device configuration, but have longer failure detection times. The more advanced resiliency levels require specific development and signal handling on the endpoint to support the Mitel model. In all cases, successful resilient behaviour cannot be assumed, but instead must be tested explicitly to ensure successful interoperation between the device and the ICP. Starting with MCD 4.0, the SIP Center of Excellence will include resiliency testing as part of all 3rd party lineside interoperability. Feature Enhancements SIP Resiliency refers to continued service from a SIP endpoint despite the endpoint being unable to reach its primary 3300 Controller. Even though the secondary remains available, when the primary is capable of resuming service for the device, the device should fail-back to its primary and re-establish service. The process of fail-over and fail-back needs to be transparent to the end user, but the time it takes for a device to fail-over and fail-back will vary widely depending on the resiliency model supported by the device. While calls can be placed to the device at any time, calls from the device cannot be handled until such time as the device recognizes its proper controller. Note SIP Resiliency, in this context, does not refer to resiliency related to SIP trunking Because the capability is as much associated with the endpoint as it is with the 3300 Controller, Mitel is not in control of all aspects of resilient behaviour. For a device to be resilient within any service model, standard programming is required on the 3300 controller and the SIP device must also accept and respond to SIP messages from any IP address, not just the address with which it has registered. This allows the 3300 Controllers to dynamically adjust the signalling path when a failure occurs, thereby routing incoming calls virtually instantly to the SIP device (the default configuration for out-of-service time for incoming calls is 45 seconds). Because of the need for capabilities in the endpoint there are multiple levels of SIP resiliency possible on the 3300 controllers. 2-6 Module 2 Mitel SIP

Mitel SIP Bronze The Bronze level of resiliency relies on DNS to provide the SIP device with an alternate address should the primary address become unreachable. There are two events which would trigger the registration with the secondary 3300 Controller. The registration refresh timer expires, and the endpoint, upon failing to register with the unavailable primary 3300 Controller, automatically initiates registration with the secondary 3300 Controller. Alternatively, if a user attempts to make a call, the endpoint will be unable to communicate with the primary 3300 Controller and will then register with the secondary in order to place the outgoing call. The time to detect loss of service from the primary 3300 Controller and fail-over to the secondary 3300 Controller is dependent on either registration refresh or user interaction. To minimize the failure detection time, the registration refresh interval can be lowered, to not less than 300 seconds. Otherwise, the resulting network traffic from set registration effects can impact total call volume scalability. If a device supports the 301 REDIRECT message on REGISTER, and also supports the concept of reconfiguring all services to the IP address the device registers with, the device will seamlessly fail-back to the primary 3300 Controller once it is available. If not, the device will need to support the priority entry for DNS entries, as well as periodically retry higher priority entries. The DNS server will need to be programmed with the Primary 3300 Controller having a higher priority in order to guarantee the Primary 3300 Controller is the primary service provider. A phone cannot be considered resilient unless it provides some form of implementation to return to the primary 3300 Controller (fail-back) once it is available Silver The Silver level of resiliency allows the primary and secondary 3300 Controller to be specified using static provisioning on the device. The device may support the use of 301 REDIRECT, which can further improve the configuration experience by allowing the device to be redirected into the MCD cluster when it s primary or secondary is reconfigured on the 3300 Controller. When the SIP device detects that the primary ICP is no longer reachable, which typically occurs via REGISTER refresh or user action (e.g. make a call), the device will attempt to communicate with the secondary. In the event of static provisioning, this may result in a 301. Gold The Gold level of resiliency encompasses all the capabilities of the silver model, with the addition that the phone is capable of using a SIP OPTION message as a heartbeat. By using a heartbeat message, the endpoint does not have to wait for the registration refresh timer to expire before detecting the failure of the primary ICP. Consequently, the registration refresh timer can be set to the recommended 60 minutes. Processing the OPTION message requires much less CPU overhead than processing the REGISTER message. Platinum The Platinum level of resiliency provides the best level of resiliency protection. The primary and secondary 3300 Controller information is provided dynamically to the phone via proprietary SIP headers. The interval for the OPTION heartbeat is also specified in this message. 2-7

Mitel SIP Workshop Summary of Levels Level Pros Cons Example of Device in this category Bronze No additional implementation required by the SIP device. Fail-back depends on device implementation. Requires frequent registration refresh interval to limit out-of-service period. Third party SIP devices that correctly support DNS- based registration. Mitel SIP endpoints; Polycom SoundPoint IP301 and IP430 Silver Ease of configuration (can provision static entry points which then issue 301 REDIRECT messages upon registration). Fail-over and fail-back still depends on the registration refresh interval. Ascom IP DECT Gold By using the OPTION message, failover detection time is improved considerably. The ICP has no control over the frequency of the OPTION message Platinum Very fast fail-over and failback detection times. Requires the phone to support a proprietary header. The phone requires no additional configuration to support resiliency. The primary and secondary ICPs can be reconfigured dynamically, and the phone automatically detects and adapts. 2-8 Module 2 Mitel SIP

Mitel SIP Lab 1 Download the latest SIP Interop Reference Step Task Expected Result/Observation 1 Open and login to Mitel OnLine 2 Select Knowledge Base from the Technical section 3 Select 3300 Integrated Communications Platform ICP from the product drop down list. 4 Search using the keywords SIP Interop 5 Select the latest SIP Interop reference document 6 Click the pdf icon representing the document The pdf documents opens. 7 Save the document to a local folder 2-9

SIP Phones 3 Objectives When you finish this module, you will be able to: Program and connect a 3 rd Party SIP Phone to MCD 4.0 Configure a Mitel IP Phone in SIP mode and connect it to a 3 rd Party SIP Server Connect a Mitel 5302 SIP phone to MCD 4.0

Mitel SIP Workshop 3-2 Module 3 SIP Phones

SIP Phones SIP Phone Support With generic SIP Phone support in the 3300 ICP, SIP endpoints can utilize the rich functionality provided by the 3300 ICP. This capability allows the creation of multi-vendor solutions based on open standards and ensures a natural fit of SIP Phones into the 3300 ICP management paradigm. SIP Phone support offers the following features: Registration and Authentication SIP devices must register with the 3300 ICP using the SIP Register method as described in RFC3261. Registration cannot occur unless the device has been configured. Authentication can be enabled on a per SIP device basis. If authentication is enabled, the SIP device will be challenged as per RFC3261 during registration and while making calls (on SIP REGISTER and INVITE methods). Billing and SMDR All calls from Generic SIP phones are represented in SMDR as calls from local extensions. Licensing Every generic SIP Phone requires a SIP user license in order to be configured. Device licenses will not be required for the SIP phone to operate. 3300 ICP Management and Configuration SIP Phones are configured within the 3300 ICP, similar to Multiline sets and Hotdesk users. Each SIP phone has a main DN associated with it. Additional lines can be programmed for the SIP Phone allowing calls for alternate DNs to be delivered to the SIP Phone. Additionally, the behavior of a given SIP device can be characterized using the SIP Device Capabilities Assignment form and then applied to SIP phones to support alternate capabilities. Support for CESID and peer trunk status is available between the 3300 ICP and third-party devices. Note Mitel does not recommend that Mitel dual mode sets be deployed in SIP mode on the 3300 ICP because in SIP mode the set provides significantly less functionality than it does in MiNET mode. 3-3

Mitel SIP Workshop Generic SIP Devices MCD 4.0 The following features are supported via defined SIP methods between 3300 ICP and Generic SIP Endpoints: Basic Call All SIP devices must be registered with the 3300 ICP, which acts as the Registrar. Once registration is completed, SIP devices can make and receive calls. Calls made to an unregistered SIP device will fail, similar to calls placed to a non-existent IP device. Calls made from a SIP device that has not registered with the 3300 ICP will receive SIP error message 404 - Not Found. Note When a handset is out of range or switched off, calls from other phones to that extension provide ringback instead of overflow tone. This behavior applies to KIRK sets and to wired SIP sets that are unplugged. Broadcast Group SIP users can participate in all different kinds of groups and have the same restrictions as MiNET sets. Call Forward The Generic SIP Phone can have Call Forwarding options programmed against its DN. The programming of call forwarding destinations, as well as the activation and deactivation of call forwarding, is performed using feature access codes. In addition to supporting the 3300 ICP forwarding behavior, SIP phones can also provide local call forwarding directly from the set using SIP REDIRECT messages defined in RFC 3261. Call Hold SIP devices can place a call on hold and music may be provided by their endpoint. Call Hold Retrieve The SIP user can retrieve the held party by dialing the feature access code for Call Retrieve. Call Park and Call Park Retrieve Calls can be parked against the user DN and retrieved by using the Call Park Retrieve feature access code. 3-4 Module 3 SIP Phones

SIP Phones Call Transfer The 3300 ICP provides RFC compliant SIP signaling support for blind (unattended) and supervised (attended) transfers. Blind transfer is based on the REFER method defined in RFC 3515. Supervised transfers rely on the "Replaces" header defined in RFC 3891, and allows the following: CANCEL transfer operation and reversion back to the original caller (in case of Mitel SIP phone, the Cancel key can be used). This must be accommodated during the ringing and answered states. RELEASE transfer resulting in the connection of the current and held party. TRADE between two calls CONFERENCE between parties involved in the transfer Calling Party Number and Name The name and number of the calling party is present to the SIP phone at setup time, if available. If the call originated on a 5ESS PRI trunk, the calling name and number may not be available. When the SIP Phone initiates a call, the name and number supplied in the FROM header will be used as the calling party name and number. Conference SIP phones support the following forms of conference: Locally hosted conference (if supported by the device - this requires enabling Multiple Line Support in the SIP Device Capabilities form) Conference bridge Conference hosted on a conference resource known to the 3300 ICP (3300 ICP hosted conference is only available if Multiple Line Support is disabled in the SIP Device Capabilities form) Connected number delivery When a SIP user makes a call to the 3300 ICP, the current number and name being alerted is displayed (wait for answer display). When the call is answered, it may not be answered by the same number that was presented to the caller in wait for answer state. If this is the case, then the answering party's name and number are displayed to the caller. For example: A calls B C has an appearance of B When B and C start ringing, the Display on A indicates B is ringing. 3-5

Mitel SIP Workshop C answers the call and the answer message contains C's name and number. Dialed Call Pickup SIP users can be members of a Pickup group. When a member in the groups is ringing and the SIP device can pick up the call using the Dialed Pickup feature access code. Directed Call Pickup SIP devices can pick up calls by using the Directed Call Pickup feature access code and the DN of the device to be picked up. Do Not Disturb SIP devices can activate Do Not Disturb by dialing the Do Not Disturb feature access code. DSS/BLF Other devices in the network can be configured to monitor the SIP device. The DSS/BLF behavior is limited when a SIP device is being monitored. On and off-hook events, and certain key presses in some cases are not monitored. Emergency Support for Generic SIP Phones This feature provides the following improvements: Caller Emergency Service Identification (CESID) when a generic SIP phone makes an emergency call. The information provided is an ID number and if programmed, a comment. If no comment is programmed for the device, then the emergency caller's telephone directory name is displayed. An SNMP event is generated to the Emergency Response Advisor when a generic SIP phone makes an emergency call. The CESID number in the SNMP event must be the default CESID number or a programmed CESID number for the generic SIP phone. Emergency Services - Location Notification is also supported on Generic SIP Phones. Feature Access Keys Generic SIP Phones: While users are logged into a SIP device, feature access keys that are available on MiNet sets will not be accessible for example, Call Forwarding, Message Indication Key, Auto Answer, and so forth. Some of these features may already be provided by the SIP phone itself. The administrator can assign any specific usage for programmable keys, since they are not defined. The key assignment shown below is an example only. 5302 SIP Phones: The 5302 SIP Phone has four programmable keys that are hard coded as CDE Speed Calls, which can be programmed in the System Administration Tool to provide the following features: 3-6 Module 3 SIP Phones

SIP Phones Key #3 Voicemail Key #4 Conference Key #5 Personal Speed Call 1 Key #6 Personal Speed Call 2 Hunt Group SIP users can be members of a hunt group and calls are presented to the SIP device as they normally would to the MiNET user. Key Line Appearance With the exception of the 5302 SIP Phone, a SIP device may have the line appearances of other devices, or broadcast groups in the system. Last Number Redial SIP devices can use Last Number redial by using the appropriate feature access code. Message Waiting Indication for Voice Mail Integration Message Waiting indications are only provided for the prime line associated with the Generic SIP phone user. Paging (Direct, Group, and Loud Speaker) A SIP phone can initiate a direct page using the Direct Page feature access code and the DN of the device to be paged. The regular COS checks apply. SIP devices can perform loudspeaker pages to specific zones or all zones by using the feature access codes. Persistent Registration of Non-Resilient SIP Devices If the 3300 ICP reboots, either due to a planned upgrade or unplanned occurrence such as a power outage, the 3300 ICP remembers the registration information for non-resilient SIP devices and automatically re-creates the registrations. This allow the SIP devices to place and receive calls without having to wait for the devices to re-register on their own at some later time. Registration of SIP Endpoints with Name This feature supports using name strings in the URI; an Internet style address, such as sip:fred@mitel.com. The feature uses the URI/Number translation table configured by the system administrator. Remote Hold Retrieve SIP devices may retrieve held calls, which reside on other devices by dialing the Remote Hold Retrieve access code. When the SIP device places a call on hard hold, others in the system can pick the call up. 3-7

Mitel SIP Workshop System Speed Call SIP devices can initiate a system speed call by dialing an abbreviated Speed Call Number. Note Always check the latest SIP Interop Reference to ensure compatibility. 3-8 Module 3 SIP Phones

SIP Phones Program Generic SIP Phones To program generic SIP telephones, use the following forms: SIP Device Capabilities Assignment To program the generic SIP device as a multiline set, enable the "Replace System based with Device based In-Call Features" field. Complete the other fields as required. Additionally, at least one of the keys of the SIP device must be configured as Multicall to its own prime DN or Key System to another DN in the Multiline Set Key Assignment form. Set this option to program as a multiline set Multiline IP Set Configuration form Complete the Device Type, Directory Number, User PIN, Confirm User PIN, Interconnect Number, and Tenant Number fields. The device type must be "Generic Device Type". Login PIN is the SIP authentication password. The username is the DN. 3-9

Mitel SIP Workshop Multiline Set Key Assignment form To program a generic SIP device as a multiline set, program additional buttons with a Line Type of "Multicall." Enter the directory number of the generic SIP device in the Button Directory Number field. Note that entering numbers others than DN of the device is not supported. 3-10 Module 3 SIP Phones

SIP Phones Multiline Set Group Assignment form (Optional) Change the ring type at each telephone where this number appears. Change the group type from "key system" to "multicall" or vice versa. Default Account Code Definition form (Optional) Create a default account code number that will appear in all SMDR records for the station. Station Service Assignment form Assign a Class of Service, Class of Restriction, and Intercept Number to the directory number of the telephone. Change the SIP Device Capabilities number, if applicable. (Optional) Assign a Default Account Code Index number. User Configuration form Use the User Configuration search field to locate the directory number that you assigned to the device. Assign a name, department, and location to the directory number. Enter the SIP Device Capabilities number, if applicable. Shared Forms Configuration Ensure that the SIP Device Capabilities form is shared. The default is "All Cluster Members." You can also restrict records from being shared by specifying the SIP Device Capabilities Numbers not to share in "Records Not Shared" portion of the form. Class of Service Options Set "Public Network Access via DPNSS" to Yes in the COS of SIP Phones to allow calls over LS trunks. Disable the Auto Camp-on Timer to allow unsupervised call transfers from the SIP device to camp on to busy station. Note If no telephone directory name is programmed for a SIP device on the 3300 ICP, the system will default to the SIP display name. For some SIP devices, the SIP display name can be provisioned. To prevent users of generic SIP devices from programming their own display name, you should always add an entry and provide a name for all of these devices in the Telephone Directory Assignment form. 3-11

Mitel SIP Workshop If you delete telephones, you must also delete the corresponding directory entries and voice mailboxes. URI/Number Translation (Optional) The URI/Number Translation form will allow a name to be substituted for a number if this is required. The outgoing number of the Mitel system is replaced with the host portion of the SIP URI. Example: 2002@academy.mitel will become sip@academy.mitel for outbound calls, similarly sip@academy.mitel will become 2002@academy.mitel for inbound calls 3-12 Module 3 SIP Phones

SIP Phones Lab 1 Connecting a 3 rd party SIP phone to a Mitel controller In this lab you will: Connect the X-lite softphone, a 3 rd party SIP softphone, to the Mitel controller Configure a Generic SIP device on the 3300 Step Task Expected Result/Observation 1 Open the 3300 System Administration tool for the controller 2 Navigate to System Configuration > Devices > SIP Device Capabilities Assignment form 3 Select number 3 in the right-hand pane Click Change 4 Fill in the form: Comment: X-Lite softphones Click Save 5 Navigate to System Configuration > Devices > User Configuration 6 In the right-hand pane, click the add button and fill in the details as follows: Last Name: Tom First Name: Gray Number: x002, where x is your student id Device Type: Generic SIP Phone SIP Device Capabilities: 3 User Pin: x002, where x is your student id Confirm User PIN: as above Desktop Admin: Remove the tick Click Save 7 You are now ready to connect the client Consult your instructor for the Class of Service and Class of Restriction values. 3-13

Mitel SIP Workshop Install and configure X-lite Step Task Expected Result/Observation 8 Open C:\Mitel Academy\SIP Workshop folder Double click X-Lite Setup.exe to begin the installation A wizard appears to guide you through the installation 9 Click Next Agree to the License Agreement, click Next Leave the default path, click Next Leave the default options selected, click Next Leave the Launch X-Lite option selected, click Finish 10 Answer any question that might be displayed until X-Lite shows the SIP Accounts dialog box 11 Click the Add button on the right and fill in the form: Display Name: Tom Gray Username: x002, where x is your student id Password: x002, where x is your student id Domain: IP Address of the controller eg 192.168.y.2, where y is you subnet id Click OK Click Close 12 You generic SIP phone is now ready to use You should see X-Lite display Registering. Then Ready Your username is: x002 3-14 Module 3 SIP Phones

SIP Phones Mitel SIP software of IP Phones Mitel has been working with SIP for years. Over that time many changes and improvements have taken place. At the time of writing this course the most recent version of SIP software was 7.2 UR1. To check if there is a newer version, login to the Mitel website and check in the Software downloads section. Note Information on updating the phone to the latest version of SIP will be covered later in the course. 3-15

Mitel SIP Workshop Changing to SIP Mode By default Mitel phones 1 arrive ready to work using the Minet protocol. To change this behaviour the phone will need to be configured. The easiest way to do this is to boot the phone whilst holding down the * and 7 buttons You will see: MODE CHANGE TO SIP Release keys=cancel then CHANGE CONFIRMED To return the set to MiNET mode, hold down * and 6 during boot up. If the phone is already loaded the Firmware Menu can be used To do this press both the up and down volume buttons at the same time, then while keeping one button down, release the second button. Still holding the remaining volume button down, dial 234 and then let go of the volume button. The following screen will be displayed: NETWORK PARAMETERS? *=YES #=NO Changing a Mitel phone mode to SIP Display What to press Display What to press NETWORK PARAMETERS? *=YES #=NO HARDWARE CONFIG? *=YES #=NO Press # Press # PHONE MODE? *=YES #=NO PROTOCOL? *=YES #=NO Press * Press * PHONE MODE: Minet *=Change #=NoChange PHONE MODE: SIP *=Change #=Accept PHONE MODE: Minet *=Minet 0=SIP #=Back Press * Press 0 STORE CHANGES? *=YES #=NO Press # Press * 1 Mitel 5302 only communicates using the SIP protocol. 3-16 Module 3 SIP Phones

SIP Phones SAVING TO NVRAM Do Not Remove Power Wait NVRAM SAVE COMPLETE Wait REBOOT NOW? *=YES #=NO Press * REBOOT NOW? Resetting Phone Wait Once the phone has rebooted, it can be configured through the web interface. 3-17

Mitel SIP Workshop Configuring a SIP mode Mitel IP Phone Quick Setup To configure a Mitel IP phone, browse to the IP address of the phone using Internet Explorer. Example: http://192.168.1.20 The default User name is admin and the password is the model of the phone. Example: If I am configuring a Mitel 5340 IP Phone, the username is admin and the password is 5340. Once Logged in you will see the Home page For a basic setup, select the Quick start menu from the left column Quick Start 3-18 Module 3 SIP Phones

SIP Phones Fill in the User ID or Extension, User Display name, SIP Authentication User name and SIP Authentication Password. Fill in the FQDN or IP Address for the SIP Proxy Server and the SIP Registrar Server. Click on Apply. The phone will automatically create a user account, which will allow the user to log in using their extension number (username) and SIP Authentication Password (password). This will enable the user to change personal setting for the phone. 3-19

Mitel SIP Workshop Detailed look at the SIP Configurations Home The Home page provides a starting point to configure the SIP phone 3-20 Module 3 SIP Phones

SIP Phones User Tools Feature Configuration TO value can be a telephone number, SIP URL, or an IP address. If left blank, calls will be forwarded to the voice mailbox programmed in the User List Configuration page. Activate this by setting it to On. Select a default message or to enter a personalized message, select Other Reason. Maximum of 20 characters. If On, a beep tone is heard by user as a reminder that a call is on hold. If On, the user hears beep tone is when a call is received. Use these settings to enable/disable display of feature messages. You may want to disable these displays to improve readability of RSS feeds or branding. HTML Player offers the ability to execute third-party applications on phones. 3-21

Mitel SIP Workshop Phone Book Names, up to 20 characters, and numbers added here are displayed in the same order when using the phonebook on the phone. Numbers and Names can also be programmed from the phone. 3-22 Module 3 SIP Phones

SIP Phones Dial by URL URL must be preceded by sip: Example sip:2001@mitel.com URL can be a maximum of 128 characters. IP Addresses can also be used. 3-23

Mitel SIP Workshop Key Programming The number of keys/pages depends on the phone model. To program a key, click the associated button. This opens a separate window for programming Key Details Select a feature to be programmed or modified on the selected key The context associates the specified key with a User ID configuration in the User List. Enter the number or address to be programmed to the key. This would also be the user being monitored by the Busy Lamp Field feature. 3-24 Module 3 SIP Phones

SIP Phones Caller ID Services This field contains a Number, SIP URL or Keyword. sip: must always precede a SIP URL. The ring tones range from low pitch (2) to high (15) 3-25

Mitel SIP Workshop Call logs The most recent call appears at the top of each log. The call information recorded may include the party name, number, SIP URL or IP address, the call duration, and the time and date of each call. Date/Time The time and date set using the Date/Time page will be overwritten if you have configured the phone to automatically synchronize with a Simple Network Time Protocol (SNTP) network time service. (Network Configuration Page) 3-26 Module 3 SIP Phones

SIP Phones Users and Passcodes This is the UserID and Password to allow access to the Web Configuration Tool. It is not the SIP Authentication user name. The default passcode for admin is the phone type (eg 5340) and for a user it is hello This is the passcode used by the phone installer in conjunction with the User ID during initial phone installation. 3-27

Mitel SIP Workshop Admin Tools Quick Start Unique name or extension number assigned to the user. Maximum of 32 characters. Name displayed on a user's phone. Maximum of 20 characters. User's SIP account name and password from your SIP Service Provider. Required only if you have a SIP Service Provider. If you do not have a SIP Service Provider, password is used to log in. The IP Address or FQDN is appended to dialed name or number. Maximum of 128 characters Used if SIP Proxy and Registry Servers are not the same. 3-28 Module 3 SIP Phones

SIP Phones User List Click the Add New button to create a new user. The User Configuration screen opens in a new window User Configuration Screen The top part is the same as the Quick Configuration SIP request and responses are sent to the outbound server. If set to Default, only the first packet is sent to the outbound server. If set to All Packets, then all packets are sent to the outbound server. Server IP address and domain name of external voice mail server. If this field is configured, the phone will connect to server using the default user name during boot-up. When Keep Alive mode is configured, if UDP is configured, Mitel SIP phones send STUN keep alive packets to the SIP Registrar (or Proxy, if Registrar field is empty). IF TCP connection is configured, phones send TCP carriage returns. 3-29

Mitel SIP Workshop Advanced Features None = no registration authentication. Basic = authentication with basic encryption. Digest = authentication with digest encryption. Time after which the phone will automatically re-register with the Registration Server. This can be used for resiliency. Set the SIP session timeout value (in seconds). If the peer does not respond within the allocated time, the session (call) will be torn down. URL of voice mail server. If this field is configured, the phone will connect to server using the default user name during boot-up. long SIP- Contact (RFC3840) When changing from one language to another, make sure that a TFTP server is running so that the new language files are downloaded. Emergency number for area and the address of server used by emergency number dialed. Enter the STUN server IP address for STUN applications to use. When enabled, phones obtain the firewall address from the specified proxy server. This enables or disables firewall NAT bypass operation. When enabled, this feature lets the phone function behind a firewall that is not SIP-aware. WAN IP Address is for the external IP Address of the router. A Globally Routable User Agent URI is a URI that routes to a specific UA instance. When enabled, the phone obtains the GRUU parameter during registration. If the server supports GRUU, the phone will include the GRUU parameter in its contact header. If GRUU is disabled, or the server does not support GRUU, the phone assumes current SIP release behavior When Hot Line Mode is On, all calls are automatically made to this address or number when the phone is taken off-hook. 3-30 Module 3 SIP Phones

SIP Phones Advanced Features continued Pressing the pound key # at the end of a dialing sequence completes Dialing. The Busy Lamp Field (BLF) feature allows you to program a key that monitors whether or not another user is on a call. The BLF Key also acts as a Speed Dial key to the monitored user s number, and as a Call Pickup key on behalf of the monitored user. Enables/disables the HTML player feature. When disabled, HTML Applications Master Filename is grayed out and is not available for user modification. For PBX-style plans, it may be necessary to dial a preceding digit (9) to deliver dial tone to the phone. Enter the preceding digit here, if required. The code used by the Server for Call Pickup Provides a method to globally customize messages shown on Line 1 of an idle SIP Phone display by linking to an RSS feed URL or an HTTP server feed or providing a Branding message (maximum 126 characters including spaces). A default URL link is supplied. To turn off RSS feed, clear this field. RSS URL provides the full internet path for the RSS Feed, Maximum buffer size of the RSS text buffer is 500 characters, remaining characters are truncated. Branding Text is identified by single quotes around the branding text to be displayed. The branding text will be displayed on Line 1 when the phone is in Idle mode. If the text is greater than the display are, then it will automatically scroll the message, Maximum branding/note length is 119 characters. Enable/disable the ability of the server to reset the phones using Notify messages. 3-31

Mitel SIP Workshop Network Configuration Required for cable access. Optional. Maximum of 128 characters. IPv4 = outgoing SIP requests use dotted format of IP address. Fqdn = outgoing SIP requests use sip:host@name.domain format for contact SIP header. The server where updates to the firmware, languages and configuration files can be downloaded using the TFTP or HTTP protocol. Server used for date/time synchronization. Date/time programmed here will overwrite date/time user may have programmed in Feature Configuration page. Changes the phone tone plan. VLAN and QoS Settings for the Phone The phone looks for configuration files on the TFTP or HTTP server when booting up. The settings in the configuration files will overwrite any manually entered settings. Point-to-Point Protocol over Ethernet. Enabled when using a DSL network only Computer-supported Telecommunications Applications (CSTA) Enable (set to On) for the 5330/5340 phones to allow Companion Applications. The CSTA Password is required for connections, enter an alpha-numeric password that you will enter to establish phone association when installing the PC Application software. Default is phone model number (for example 5340). 3-32 Module 3 SIP Phones

SIP Phones Dial Plan When activated, this feature forces all dialed digits to use the inter-digit timer specified in the timer. The Dial soft key or # will be disabled, and the entered digits will be dialed automatically after the timer has expired. Enter the number of digits expected to follow the partial number Note that the Followed By field must be set to 0 for the.t parameter to work. The timer monitors the keypad for a pause in digit entry. When the timeout interval is exceeded, the digits are optionally modified, then sent to the SIP registration server or alternate server. Enter a partial or complete digit string (2 16 digits) for use as a template to test dialled digits. If the value of one or more digits is variable, then enter the wildcard x in place of each variable digit. Use the square brackets [ ] to specify a set of digits, any one of which can match a dialled digit. For example, [123] would be a positive match if the user dials 1, 2 or 3, but would fail to match 4. The.T timer parameter (which must be placed at the end of a digit string) causes the phone to accept an arbitrary number of digits from the user. Once the user has finished entering digits (signalled by no new digits for a period equal to the timer value), the digit string may have digit manipulation performed on it. The number is sent to the SIP registration server without requiring the user to press the Dial soft key or #. Specify the IP address or domain name of another SIP registration server (an alternate route) that calls meeting the criteria specified will be automatically routed to. If left blank, the digits will be routed to the default SIP registration server. Can be used to specify suffix digits that are added to the digit string before the string is sent to the SIP registration server for dialing. The digit string can be: a number user name SIP URL SIP URL priority IP address The phone tests digit strings by examining the top row first. If a match is not found in the first row, then the phone proceeds down the table until either a match is found or all the rules are exhausted. If a match is not found in the table, then the phone will send the digits exactly as they were entered to your SIP registration server for dialing. Effective use of Dial Plan rules can eliminate the need for the user to terminate many common calls, and can simplify calls to special services such as discount long distance carriers. 3-33

Mitel SIP Workshop Ethernet Configuration Auto setting allows the phone and PC to automatically negotiate the Ethernet speed and duplex. If the PC does not support automatic negotiation, then manually select a compatible setting. Protocols Disabling HTTP access prevents future web-based sessions with the phone (i.e. prevents access to the Web Configuration Tool). If you need access to the tool, then enable HTTP through the phone's Settings menu interface. HTTPS allows for encrypted sessions. Deactivating these protocols prevents communications with the phone via TFTP, Telnet or SNMP. Activating this protocol encrypts media protocols. 3-34 Module 3 SIP Phones

SIP Phones Users and Passcodes This is the UserID and Password to allow access to the Web Configuration Tool. It is not the SIP Authentication user name. The default passcode for admin is the phone type (eg 5340) and for a user it is hello This is the passcode used by the phone installer in conjunction with the User ID during initial phone installation. 3-35

Mitel SIP Workshop Media Configuration The Media Configuration page allows you to establish the audio parameters used by the phone when communicating with other devices. G729A = 8:1 compression codec Three-way conference calls support only one G729A stream. Frame size in milliseconds used by G711 or G729 codecs. Increased in 10ms increments. Used for DTMF tone generation (RFC2833). Outband used for RTP DTMF, default payload type is 101. These fields specify the UDP port range used by the phone. Registration The Registration page allows you to: Determine whether the phone is registered with a SIP server and, if so, for how long. View the current status of the registration process. For example, "retrying" or "renewing". Manually initiate a registration request. The registration process is automatic. The Registration page is provided for troubleshooting purposes. 3-36 Module 3 SIP Phones

SIP Phones Firmware Update The main purpose of the boot load is to start the main load, and re-install the main load. Displays version of current main and boot firmware loads. Select the timer type for automatic upgrades. Absolute Time - choose an actual time of day for the polling to take place Polling Interval - choose an interval from 30-1440 minutes Turn HTTP or TFTP firmware upgrade to On or Off, or set the upgrade to occur automatically. Off - firmware upgrades cannot occur On - you are prompted for a firmware upgrade if the firmware on the server differs from the firmware on the phone Auto - automatic upgrades if the firmware on server differs from firmware on the phone Phone must be idle for an automatic upgrade to occur. A manual upgrade overrides all current firmware settings except for your Service Provider's TFTP server settings. If the download type is HTTP or TFTP, enter the server URL from which the new firmware will be installed. Directory paths and IP addresses are also supported. 3-37

Mitel SIP Workshop Configuration Upload/Download The Configuration Upload/Download page allows you to load, restore, or backup and save phone configuration files by loading the files between the phone and the PC. 3-38 Module 3 SIP Phones

SIP Phones Support Help If you select the Help hyperlink, you will be redirected to edocs.mitel.com and have access to the resources relating to the phone type. 3-39

Mitel SIP Workshop Lab 2 Configure a Mitel Phone in SIP mode In this lab you will: Configure a Mitel phone in SIP mode Connect the Mitel phone to a 3 rd Party SIP Server Configure and Connect a Mitel 5224 phone in SIP Mode Step Task Expected Result/Observation 1 Hold down the * and 7 keys and reboot the phone Keep the keys held down until you are prompted to release them Leave the phone to boot up If the keys are released too early, the phone remains in Minet mode 2 Using Internet Explorer, browse to the IP address of the phone Use the following to authenticate: Username: admin Password: 5224 The IP Address can be obtained from the System Configuration > IP Network Configuration > DHCP > DHCP Lease Viewer form on the 3300 The default password is the model of the phone. 3 Select the Quick Start option, in the left-hand pane under Admin Tools 4 Fill in the Quick Start form: Note: x is your student id User Details User ID: 500x User Display Name: Your Name SIP authentication User Name: 500x SIP Authentication Password: 500x SIP Servers SIP Proxy Server: IP Address from Instructor Port: 5060 Scheme: UDP SIP Registrar Server: IP Address from Instructor Port: 5060 Scheme: UDP Click Apply Click OK 3-40 Module 3 SIP Phones

SIP Phones 5 The phone is now ready to use on a 3 rd Party system Testing the Mitel 5224 in SIP mode Step Task Expected Result/Observation 6 Use the phone to call another student who has completed the previous exercise 3-41

Mitel SIP Workshop Mitel 5302 and 5304 IP Phones The 5302 SIP Phone is an entry-level, two-line, dual-port telephone that interfaces directly to the 3300 ICP using SIP protocol. The phone provides three fixed feature keys (Hold, Redial, Transfer) four programmable keys, and supports G.711 and G.729 compression. The 5302 IP Phone has a speaker for paging purposes; LED s and tones are used to provide feedback. The Mitel 5304 IP Phone is a two-line, dual port telephone that provides voice communication over an IP network. It has a back-lit liquid crystal display (LCD) screen. The 5304 IP Phone offers 8 programmable keys for one-touch feature access and one prime line key. The personal key on the bottom is always your Prime Line. The 5304 IP Phone supports compression, and is designed for users requiring access to basic telephony and messaging services. It has 2x20 back-lit LCD and features 8 programmable keys. The 5304 IP Phone supports Mitel Call Control (MiNet) protocols and session initiated protocols (SIP). 3-42 Module 3 SIP Phones

SIP Phones Configuring a Mitel 5302 IP Phone Open the User Configuration form. Select the Add button and fill in the form using 5302 as the device type. The user PIN can be 0-8 digits long. The 5302 has no resiliency To give the phone a multiline presence, use the Multiline Set Key Assignment form and add in a muliticall for the second button using the original number. 3-43

Mitel SIP Workshop The DN of the 5302 Now boot the 5302. Use the following table to ensure that the phone is registered with the controller 5302 Boot Process Step Indication MWI Line 1 Line 2 Tone LED LED LED 1 Boot-loader start-up On On On - 2 Software (Kernel) start-up On On On - 3* A Directory Number (DN), and optional PIN, must be entered. If the DN/PIN is already stored, then proceed to step 5. 4* Enter the DN (1-7 digits) and then press #, enter the PIN (0-8 digits) and the press # 5 Bootstrapping will continue within 5 seconds, using DHCP. To reset the DN/PIN, and revert to factory defaults, press * key. The phone proceeds to step 3 and reverts to factory defaults after the DN/PIN has been entered. Flash Flash Flash - Flash Flash Flash DTMF feedback is played for valid alphanumeric key presses. - Flash - - 6 Network Connection tested - - - - 7 Perform DHCP negotiation - Flash Flash - 3-44 Module 3 SIP Phones

SIP Phones Step Indication 8 Register with the call server. If there is no call server stored in the phone, extract the call server address from option 125. If there is no call server supplied by DHCP and there is no call server stored in the phone, the phone restarts. 9 If redirected to different call server, attempt to register with the contact address specified in the redirect message. Once successfully registered the call server is stored. 10 Download the local network profile and device profile from the call server. 11* If the current firmware version differs from the device profile firmware version, download the firmware from the TFTP server else proceed to step 13. Disabling firmware upgrades will cause this step to be skipped. MWI LED Line 1 LED Line 2 LED Tone - - - - - - - - - - - - On On - - 12* Store firmware, and restart. On - On - 13 The phone subscribes to the Voicemail Server and is now operational (proceed to step 15) - - - - 14 Registration failed Flash - Flash - 15 If handset is lifted, selected line LED is on. - Selected line LED On - * These steps are not necessarily carried out for each boot sequence. Note Unlike generic SIP Phones, the 5302 IP Phone cannot provision the display name itself and defaults to "username," which is typically the DN of the device. If a name is required then add an entry and provide a name in the Telephone Directory Assignment form. 3-45

Mitel SIP Workshop Configuring a Mitel 5304 IP phone To program a 5304 IP Phone, Use the User Configuration form Select Add and fill in the details. The 5304 can be resilient A MAC Address can be assigned if required If required, the 8 programmable keys can be configured 3-46 Module 3 SIP Phones

SIP Phones The phone is added to the controller by entering a PIN and presses the hook switch. 3-47

Mitel SIP Workshop Lab 3 Connecting a 5302 to a Mitel Controller In this lab you will: Program and connect a Mitel 5302 SIP Phone to a Controller Test the functionality of the 5302 Configure a Mitel 5302 SIP phone on the 3300 ICP Step Task Expected Result/Observation 7 Open the 3300 System Administration tool for the controller 8 Navigate to System Configuration > Devices > SIP Device Capabilities Assignment form 9 Select number 2 in the right-hand pane Click Change 10 Fill in the form: Comment: Mitel 5302 Phones Click Save 11 Navigate to System Configuration > Devices > User Configuration 12 In the right-hand pane, click the add button and fill in the details as follows: Last Name: Ann First Name: Greene Number: x001, where x is your student id Device Type: Generic SIP Phone SIP Device Capabilities: 2 User Pin: x001, where x is your student id Confirm User PIN: as above Desktop Admin: Remove the tick Click Save Consult your instructor for the Class of Service and Class of Restriction values. Configure a Mitel 5302 SIP phone Step Task Expected Result/Observation 13 Plug in the Mitel 5302 SIP phone 3-48 Module 3 SIP Phones

SIP Phones 14 Wait for the MWI and LED 1 and 2 to all be in a flashing state Lift the handset and dial x 0 0 1 #, where x is your student id (this is your DN) then dial x 0 0 1 # (This is your password) Replace the handset and wait for all the lights to go out If the boot up process, continues past this stage, see the 5302 setup table, step 5 to reset the PIN 15 The phone is now ready to be used 3-49

SIP Trunks 4 Objectives When you finish this module, you will be able to: Configure a SIP Trunk to an Internet Telephony Service Provider (ITSP) Configure a SIP trunk to another Mitel 3300 ICP Configure a SIP Trunk to a 3 rd Party Application

Mitel SIP Workshop 4-2 Module 4 SIP Trunks

SIP Trunks SIP Trunks Service Providers offer SIP trunks that provide flexible and cost-effective WAN solutions for the 3300 ICP. SIP trunks allow the 3300 ICP to connect to the Service Provider through the SIP protocol over the IP network. The SIP Trunking solution provides support for the following features: 9-1-1 SIP Trunking supports 9-1-1 emergency service. The SIP Service Provider can be chosen as the outgoing emergency route. Ensure that the CESID information is programmed. Billing and SMDR The Service Provider bills calls based on the peer connection to the 3300 ICP. The 3300 ICP records are created with a special SMDR tag entered in the SIP Peer Profile form. An SMDR tag can be enabled in the SIP Peer Profile form for outgoing and incoming calls. DISA Call For Release 7.0, one restriction exists on a SIP-to-DISA call. During the call, the user cannot enter # followed by a remote number. DNS Support Communication between SIP Service Providers and the 3300 ICP can be configured to use either Fully Qualified Domain Names (FQDN) or IP Addresses. The Network Element Assignment form provides for the configuration of both for the SIP Peer. On the SIP Peer Profile form, the user can program the Local address as either a FQDN or an IP Address. FAX The SIP Trunking for Service Provider configuration supports FAX calls over G.711. Attempts to switch to T.38 are rejected, and the call continues as G.711. It is recommended that FAX machines be connected locally or through TDM to the 3300 ICP that is connected to the Service Provider through SIP trunks. The SIP Satellite Office Solution configuration supports FAX calls over T.38. For more information on this configuration, see SIP Satellite Office. FAX routing can be configured through third party gateways using either prefix routing or COR routing. Prefix routing chooses a route based on a dialed prefix. COR routing chooses a route based on the COR group restriction on a route. If a particular device belongs to a COR group, it may be restricted from dialing out a particular route and then dials out the next route available in the route list. FAX tone detection is used to disable the adaptive jitter buffer and echo cancellation to improve the reliability of FAX transmissions over IP networks. FAX tones can be detected in one direction (the TDM side) on the following types of calls: IP trunk to TDM, and SIP trunk to TDM. The Class of Service Option "Campon Tone Security/FAX Machine" is used to limit the codec selections to G.711 for Fax calls. 4-3

Mitel SIP Workshop Malicious Call Trace For incoming SIP calls that are tagged for Malicious Call, the 3300 records the Media IP address and port used remotely. As well, the SIP signalling information is captured. This information cannot be sent to the SIP Service Provider, but the information is recorded if needed. Note Malicious Call SMDR records are logged on the 3300 ICP. SIP endpoints cannot invoke Malicious Call Trace, but is it recommended that SMDR be enabled for SIP devices and gateways. NAT Keepalive This option can be configured in the SIP Peer Profile form and may be necessary for the connection to the Service Providers. It allows periodic packets of audio when a one-way connection is detected to keep the NAT firewall open. Register Each SIP Trunk can register with a registrar. The registrar is assigned using the Network Element Assignment form. Note For a complete list of RFC s that the Mitel support, please refer to the online Help SIP Peer Status The system administrator can select from Always Active, Disabled, or Auto-Detect/Normal for SIP Trunks using the Network Element Assignment form. 4-4 Module 4 SIP Trunks

SIP Trunks Phone Features over a SIP Trunk The following features should be available of a SIP Trunk: Basic Call When a call originates from the 3300 ICP to the SIP service provider, it terminates on a SIP route. This SIP route is directed to the SIP Peer and the digits dialed are presented to the service provider. When a call originates from the SIP service provider to the 3300 ICP, it is treated as an external PSTN-type call. The calling line category of the incoming call is set to CLC_ISDN. This allows for external treatment of the call for features such as call forwarding, call rerouting, camp-on, recall, and so forth. Calling number and name delivery From Service Provider - the calling number and name are based on what is available in the SIP header: first from P-Asserted-Identity, second from P-Preferred Identity, and third from the From header line. In some cases, this is not the information of the calling party, but the Parent number from the remote PBX/Softswitch placing the call. The P-Preferred Identity is only available in a back-to-back connection with another SIP 3300 ICP. To Service Provider - The calling party information may be configured on a per peer basis. By default, the caller's name and number are delivered but can be suppressed using the SIP Peer Profile form. Prior to programming calling party information, CPN Substitution is configured. For SIP, the DID Range form is used as part of the SIP Peer Profile form. In order to configure CPN Substitution, the user configures a default Calling Line ID and zero or more DID ranges. Privacy Privacy on Outgoing Calls - the existing privacy mechanisms for the 3300 ICP can be used: feature access code for Name Suppression on Outgoing Trunk Call, and the Telephone Directory Assignment form using the Private field. The SIP Peer Profile form has an option for Calling Line ID Restriction. This option replaces the calling party name and number with anonymous for display. As well, a parent account may be in used based on the SIP Peer Profile configuration. If the parent account is used, it prevents the calling name and number from being displayed. Privacy on Incoming Calls - incoming calls can contain a privacy header that prevents the caller name and number from being presented to untrusted networks or devices. Call Hold When a 3300 ICP user places a SIP Trunk on hold, the 3300 ICP provides Music on Hold, if programmed. The feature behaves similarly to PRI calls. Call Transfer Call Transfer is allowed on SIP Trunks and the functionality is the same as PRI trunking. 4-5

Mitel SIP Workshop Call Diversion Call Diversion is not available for the SIP service provider. If the service provider sends a redirect request to the 3300 ICP, the call is cleared and the user receives re-order tone. Call Forward Calls on SIP Trunks are re-directed when the Call Forwarding feature is enabled on the 3300 ICP destination. The feature behaves similarly to PRI calls. Incoming Call Handling The SIP Service Provider may use a TEL URI calling scheme to the 3300 ICP. The SIP Peer Profile for Incoming DID form contains the mapping of the dialed digits to the SIP Peer. This mapping allows the correct policy to be applied to the incoming call. Different policies can be assigned based on the digits dialed from the Service Provider. Direct Inward Dialing The SIP Service Provider supports Direct Inward Dialing (DID) to the 3300 ICP. Use the SIP Peer Profile for Incoming DID form to configure DID dialing. The incoming digits are limited to seven digits or less when translated by call control, and can terminate at stations, hunt groups, ACD paths, and speedcall. This feature operates the same as DID calling on PRI trunks. Trunk service options to insert or absorb digits work the same as they do on other trunk types. 4-6 Module 4 SIP Trunks

SIP Trunks Programming a SIP Trunk Before you begin Ensure you have the following information: the IP address or Fully Qualified Domain Name (FQDN) of the SIP peer the transport type and the port number for the SIP peer unless it's obtained automatically through DNS the FQDN or IP address of the Registrar (if applicable) the transport type and the port number for the Registrar unless it's obtained automatically through DNS (if applicable) the FQDN or IP address of the outbound proxy server, the transport type, and port number, if an outbound proxy server is being used the FQDN or IP address of the external proxy, the transport type, and port number, if an external SIP proxy is being used the DID numbers assigned to the 3300 ICP by the Service Provider the prefix that is being added to all incoming calls for a gateway the main DID number for registration (if applicable) 4-7

Mitel SIP Workshop Programming forms To program a SIP Trunk, complete the following forms. Make sure you have the details provided by the Service Provider to complete the forms appropriately. License and Option Selection Enter the total number of licenses in the SIP Trunk Licenses field. 4-8 Module 4 SIP Trunks

SIP Trunks Network Element Assignment Dependant on your connection, you may need to program: a network element for the local switch a network element for each SIP peer, gateway, or Service Provider a network element for the Outbound Proxy if one exists in your network Note For detailed information about the SIP Peer and Outbound Proxy Network Elements, please see Appendix A 4-9

Mitel SIP Workshop System IP Port Assignment Change the SIP UDP, TCP, or TLS port number if it is different from the default value. DID Ranges for CPN Substitution To set up the CPN Substitution table for outbound calls, enter a DID number or a range of DID numbers assigned in the system. Then enter the corresponding CPN substitution number that will be delivered for that range. 4-10 Module 4 SIP Trunks

SIP Trunks SIP Peer Profile For each SIP Peer, enter the required information in this form. 4-11

Mitel SIP Workshop SIP Peer Profile continued Note For detailed information about the SIP Peer Profile, please see Appendix A 4-12 Module 4 SIP Trunks

SIP Trunks SIP Peer Profile Assignment for Incoming DID To associate a range of telephone numbers assigned by a SIP Service Provider to a particular SIP Peer, enter the required information in this form. Trunk Service Assignment: Configure the trunk as non-dial in or dial-in: update the Non-Dial-In Trunks Answer Point field for the incoming calls. strip the number of leading digits in Dial-In Trunks Incoming Digit Modification Absorb field add the appropriate number of digits in Dial-In Trunks Incoming Digit Modification Insert field. Route Assignment. ARS continues to be provisioned in the traditional manner by assigning a route to a digit or digit string. Complete the following fields in this form: select SIP Trunk from the pull-down list in Routing Medium. select a SIP Peer Profile label from the SIP Peer Profile pull-down list. enter a Class of Restriction group number in COR Group Number. enter any required digits in Digits Before Outpulsing. (If this field is left blank, digits will be sent out as "Enbloc".) Class of Service Options Assignment Enable the Public Network Access via DPNSS field in the class of service for all devices that make outgoing calls through SIP trunks, PRI trunks, LS trunks, and so forth that are connected to SIP Trunks. 4-13

Mitel SIP Workshop Conditions and Dependencies of a SIP Trunk The Class of Service for all devices connected to SIP Trunks must include the option Public Network Access via DPNSS set in the Class of Service Options Assignment form. If FQDN's are used, ensure that the Domain Name Server (DNS) in the System IP Configuration is programmed and the DNS Server is available. The route must be deleted from the Route Assignment form before deleting the SIP Peer Profile. A SIP Peer Profile must be deleted from the SIP Peer Profile form before deleting the SIP Peer from the Network Elements form. Before deleting a SIP Peer Profile, all Outgoing DID Ranges for a SIP Peer Profile must be deleted. Before clearing an Index from the DID Ranges for CPN Substitution form, all entries in the Outgoing DID Ranges form using the index must be removed. A SIP Peer profile label used by the Route Assignment form or the SIP Peer Profile by Incoming DID form must be removed before removing the SIP Peer Profile. If Challenge-based incoming call authentication is required, set the authentication realm string on the remote peer to the network element name of the local network element. If this name is not specified, set the authentication string to the domain name programmed in the System IP Configuration form. Some SIP authentication implementations only require the username and password. If Challenge-based outgoing call authentication is required, the 3300 ICP does not require provisioning of the authentication realm string. Only the username and password are required. 4-14 Module 4 SIP Trunks

SIP Trunks Configuring a SIP Trunk to a Third Party Provider Mitel is able to connect to an ever increasing variety of SIP Providers. Each Provider may have a different set of values that need to be used. In general these are the setting that would be configured, however always check the latest SIP Interop for which providers are compatible: Class of Service Make sure Public Network Access Network Element is set to Yes Network Element Assign a name that would easily identify the Service Provider Use the details provided by the Service Provider. If using an FQDN, make sure DNS is configured. Type: ping <FQDN> from the maintenance port to test it. It should normally be UDP port 5060 4-15

Mitel SIP Workshop SIP Peer Profile Once again, choose a name that is recognised easily The DDI range for the Provider may be required here Use an IP Address, unless you have DNS configured for external access. 4-16 Module 4 SIP Trunks

SIP Trunks SIP Peer Profile continued Your username for authenticating calls may be your main number. I could also be a more generic one like your company name. If you want to authenticate incoming calls ensure that the Service Provider will send the correct details. Common Changes to make from the defaults. SIP Peer Profile Assignment by Incoming DID These will be the numbers you are going to receive. Make sure they match exactly the numbers the Service Provider sends, otherwise the calls will be rejected because the server doesn t know which profile to use. Finally configure ARS appropriately 4-17

Mitel SIP Workshop Tools and Equipment To get hold of the latest information about setting up Trunks, Please search the Knowledge Base on Mitel OnLine. Use SIP as the keyword in your search 4-18 Module 4 SIP Trunks

SIP Trunks Lab 1 Configuring External SIP Trunks In this lab you will: Connect a Mitel Controller to external SIP Trunks Identify and record needed information Step Task Expected Result/Observation 1 Open the System Administration tool for your controller 2 Navigate to System Configuration > System Capacity > License and Option Selection 3 Record the number of trunks you have available on the controller: SIP Trunk Licenses: 4 Record the details needed for the SIP Trunk Provider: FQDN or IP Address: Protocol and Port: Username: Password: Inbound Numbers: The Instructor will have these available DNS must be configured if using FQDN Usually UDP 5060 Used to identify how to deal with incoming numbers Configure Mitel Controller Step Task Expected Result/Observation 5 Navigate to System Configuration > Voice Network Elements > Network Element Assignment 4-19

Mitel SIP Workshop 6 Click the Add button and fill in the form: Name: SipProvid Type: Other FQDN of IP Address: <See step 4> SIP Peer: Enable this option SIP Peer Specific SIP Peer Transport: UDP SIP Peer Port: 5060 Click Save 7 Navigate to System Configuration > Trunks > Class of Service Ensure COS 90 has not been used select it then click change Any COS can be used for SIP. If you use a different COS record the number here. 8 Complete the form by changing the following from the default values: Public Network Access via DPNSS: Yes 9 Ensure that all devices making outgoing call through the trunk have the following Class of Service options set: Mandatory Public Network Access via DPNSS: Yes Optional Busy Overide Security: Yes Campon Tone Security/Fax Machine: Yes 10 Navigate to System Configuration > Trunks > IP Networking\XNET > Trunk Service Assignment Ensure TSA 9 has not been used select it then click change 11 Complete the form: Class of Service: 90 Dial In Trunks Incoming Digit Modification Absorb: 0 Comment: SIP Provid 12 Navigate to System Configuration > Trunks > IP Networking\XNET > SIP Peer Profile Click Add 4-20 Module 4 SIP Trunks

SIP Trunks 13 Fill in the form: SIP Peer Profile Label: SipProvid Network Element: SipProvid Local Account information Registration User Name: <See Step 4> Address Type: IP Address Calling Line ID Default CPN: <Your main SIP line> Policies Trunk Service: 9 Maximum Simultaneous Calls: 4 Enable Mitel Proprietary SDP: No Force sending SDP in initial Invite message: Yes Prevent the use of IP Address 0.0.0.0 in SDP message: Yes Authentication User Name: Sysx, x is system id Password: password Confirm Password: password Authentication Option for Incoming Calls: Validate Address FQDN could be used if DNS is configured 14 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Digit Modification Ensure Digit Mod 9 has not been used select it then click change 15 Complete the form: Number of Digits to Absorb: 1 Click Save 16 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Route Assignment Ensure Route Number 9 has not been used, select it then click change 17 Fill in the form: Routing Medium: SIP Trunk SIP Peer Profile: SipProvid Digit Modification Number: 9 Click Save 4-21

Mitel SIP Workshop 18 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > ARS Digits Dialed Assignment Click the Add button If a route for external numbers already exists, then modify it to use the new SIP provider 19 Fill in the form: Digits Dialed: 9 Number of Digits to follow: 11 Termination Type: Route Termination Number: 9 Click Save Test the External SIP Trunk Step Task Expected Result/Observation 20 Using each of the phones in turn, test the External SIP trunk by dialing external numbers Eg 902073511100 4-22 Module 4 SIP Trunks

SIP Trunks Configuring a Mitel to Mitel SIP Trunk Although SIP Trunks are supported between Mitel systems, it is recommended that an IP Trunk is used for better functionality. When configuring the Trunk, use IP Addresses on the Network Elements, and the defaults provided on the SIP Peer Profile will work. Changes may need to be made, as with external trunks, to allow for multiple device types and networking configurations. Make sure the Authentication is configured to increase security. 4-23

Mitel SIP Workshop Lab 2 Connecting two Mitel controllers In this lab you will: Connect two Mitel controllers via SIP trunks Note To complete this lab, you will need to work with another group. Both sides will need to be setup before the trunk will work correctly. Identify and record needed information Step Task Expected Result/Observation 1 Open the System Administration tool for your controller 2 Navigate to System Configuration > System Capacity > License and Option Selection 3 Record the number of trunks you have available on the controller: SIP Trunk Licenses: 4 Record the details needed for the SIP Trunk to the Mitel system: FQDN or IP Address: Protocol and Port: Username: Password: Inbound Numbers: The Instructor will have these available DNS must be configured if using FQDN Usually UDP 5060 Used to identify how to deal with incoming numbers Configure Mitel Controller Step Task Expected Result/Observation 5 Navigate to System Configuration > Voice Network Elements > Network Element Assignment 4-24 Module 4 SIP Trunks

SIP Trunks 6 Click the Add button and fill in the form: Name: MitelSIP Type: Other FQDN of IP Address: <See step 4> SIP Peer: Enable this option SIP Peer Specific SIP Peer Transport: UDP SIP Peer Port: 5060 Click Save 7 Navigate to System Configuration > Trunks > Class of Service Ensure COS 89 has not been used select it then click change Any COS can be used for SIP. If you use a different COS record the number here. 8 Complete the form by changing the following from the default values: Public Network Access via DPNSS: Yes 9 Ensure that all devices making outgoing call through the trunk have the following Class of Service options set: Mandatory Public Network Access via DPNSS: Yes Optional Busy Overide Security: Yes Campon Tone Security/Fax Machine: Yes 10 Navigate to System Configuration > Trunks > IP Networking\XNET > Trunk Service Assignment Ensure TSA 10 has not been used select it then click change 11 Complete the form: Class of Service: 89 Dial In Trunks Incoming Digit Modification Absorb: 0 Comment: Mitel SIP 12 Navigate to System Configuration > Trunks > IP Networking\XNET > SIP Peer Profile Click Add 4-25

Mitel SIP Workshop 13 Fill in the form: SIP Peer Profile Label: MitelSip Network Element: MitelSip Local Account information Registration User Name: <See Step 4> Address Type: IP Address Calling Line ID Default CPN: <Your main SIP line> Policies Trunk Service: 10 Maximum Simultaneous Calls: 4 Authentication User Name: Mitel Password: password Confirm Password: password Authentication Option for Incoming Calls: Challenge-based Authentication FQDN could be used if DNS is configured Dependant on licenses 14 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Digit Modification Ensure Digit Mod 10 has not been used select it then click change 15 Complete the form: Number of Digits to Absorb: 0 Click Save 16 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Route Assignment Ensure Route Number 10 has not been used, select it then click change 17 Fill in the form: Routing Medium: SIP Trunk SIP Peer Profile: MitelSip Digit Modification Number: 10 Click Save 4-26 Module 4 SIP Trunks

SIP Trunks 18 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > ARS Digits Dialed Assignment Click the Add button If a route for external numbers already exists, then modify it to use the new SIP provider 19 Fill in the form: Digits Dialed: x where x is lab partner student id Number of Digits to follow: 3 Termination Type: Route Termination Number: 10 Click Save Test the External SIP Trunk Step Task Expected Result/Observation 20 Using each of the phones in turn, test the SIP trunk by dialing the other systems extentions Eg x001, where x is the lab partner student id 4-27

Mitel SIP Workshop Connecting MCD 4.0 to SIP Applications SIP Applications work in a similar way to External trunks. 4-28 Module 4 SIP Trunks

SIP Trunks Lab 3 Connecting to Applications via SIP trunks In this lab you will: Connect to a 3 rd Party Application via a SIP trunk Note To complete this lab you will need to have the Microsoft Exchange UM Virtual Machine running. This will significantly slow down the operation of your computer. Please allow for 15 minutes for the machine to boot, and then time for the SIP trunk to be established. Start the Virtual Machine Step Task Expected Result/Observation 1 Browse to C:\Mitel Academy\SIP Workshop and Double click the Exchange UM.vmc icon From the Virtual PC window, Select Exchange UM in the list of Virtual PC s and click Start This could take a few minutes to boot 2 Logon to the Server, using Alt Gr + Del: Username: Administrator Password: Pa$$w0rd 3 Change the IP Address to one within your subnet Eg 192.168.y.200, where y is your subnet id 4 Test connectivity with the virtual PC from the controller: Navigate to Maintenance and Diagnostics > Manitenance Commands > All Tracert rtc 192.168.y.200, where y is subnet id Click Submit The System Response should show the time taken for a response 4-29

Mitel SIP Workshop Identify and record needed information Step Task Expected Result/Observation 5 Open the System Administration tool for your controller 6 Navigate to System Configuration > System Capacity > License and Option Selection 7 Record the number of trunks you have available on the controller: SIP Trunk Licenses: 8 Record the details needed for the SIP Trunk to the Mitel system: FQDN or IP Address: <192.168.y.200> Protocol and Port: <TCP 5060> Username: Password: Inbound Numbers: The Instructor will have these available DNS must be configured if using FQDN Usually TCP 5060 Used to identify how to deal with incoming numbers Configure Mitel Controller Step Task Expected Result/Observation 9 Navigate to System Configuration > Voice Network Elements > Network Element Assignment 10 Click the Add button and fill in the form: Name: ExchangeUM Type: Other FQDN of IP Address: <See step 8> SIP Peer: Enable this option SIP Peer Specific SIP Peer Transport: TCP SIP Peer Port: 5060 Click Save 11 Navigate to System Configuration > Trunks > Class of Service Ensure COS 88 has not been used select it then click change Any COS can be used for SIP. If you use a different COS record the number here. 4-30 Module 4 SIP Trunks

SIP Trunks 12 Complete the form by changing the following from the default values: Comment: Exchange Public Network Access via DPNSS: Yes 13 Ensure that all devices making outgoing call through the trunk have the following Class of Service options set: Mandatory Public Network Access via DPNSS: Yes Optional Busy Overide Security: Yes Campon Tone Security/Fax Machine: Yes 14 Navigate to System Configuration > Trunks > IP Networking\XNET > Trunk Service Assignment Ensure TSA 11 has not been used select it then click change 15 Complete the form: Class of Service: 88 Dial In Trunks Incoming Digit Modification Absorb: 0 Comment: Exchange 16 Navigate to System Configuration > Trunks > IP Networking\XNET > SIP Peer Profile Click Add 17 Fill in the form: SIP Peer Profile Label: Exchange Network Element: Exchange Local Account information Registration User Name: <See Step 8> Address Type: IP Address Policies Trunk Service: 11 Maximum Simultaneous Calls: 4 Enable Mitel Proprietary SDP: No Force sending SDP in initial Invite message: Yes Prevent the use of IP Address 0.0.0.0 in SDP message: Yes FQDN could be used if DNS is configured Dependant on licenses 4-31

Mitel SIP Workshop 18 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Digit Modification Ensure Digit Mod 11 has not been used select it then click change 19 Complete the form: Number of Digits to Absorb: 0 Click Save 20 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > Route Assignment Ensure Route Number 11 has not been used, select it then click change 21 Fill in the form: Routing Medium: SIP Trunk SIP Peer Profile: Exchange Digit Modification Number: 11 Click Save 22 Navigate to System Configuration > Trunks > Automatic Route Selection (ARS) > ARS Digits Dialed Assignment Click the Add button If a route for external numbers already exists, then modify it to use the new SIP provider 23 Fill in the form: Digits Dialed: 8600 Number of Digits to follow: 0 Termination Type: Route Termination Number: 11 Click Save Configure the Microsoft Exchange Server Step Task Expected Result/Observation 4-32 Module 4 SIP Trunks

SIP Trunks 1 Navigate to Network Connections on the Windows Server Start > Control Panel > Network Connections Right click the Local Area Connection, choose Properties Select Internet Protocol (TCP/IP) and click the Properties button Modify the Setting to match your subnet eg, where subnet id is y IP Address: 192.168.y.200 Subnet Mask: 255.255.255.0 Default Gateway: 192.168.y.254 DNS: 192.168.y.200 Click OK, OK 2 Open the Exchange Management Console from the Start menu 3 Expand the Exchange Organization in the left-hand pane and select Unified Messaging 4 Select UM IP Gateway in the right-hand pane, right-click the Mitel controller and choose Properties Modify the IP Address to be your controller Eg, 192.168.y.2 Click OK 5 To test full functionality you will need to create user accounts and mailboxes Please see your instructor for additional information Test the External SIP Trunk Step Task Expected Result/Observation 6 Using each of the phones in turn, test the SIP trunk by dialing the 8500 4-33

Maintaining and Troubleshooting 5 Objectives When you finish this module, you will be able to: Update the Firmware on a Mitel IP Phone Setup a Network Trace Capture Network Traffic and Identify SIP packets Use SIP commands on the 3300 ICP

Mitel SIP Workshop Updating the SIP Firmware You can update the phone's firmware manually or automatically. To ensure that you are following the correct upgrade procedure, please refer to the latest product Release Notes. Note: To view the current firmware versions, see Viewing the Firmware Versions. Manually Updating the Firmware To manually upgrade the phone's firmware, download the new firmware from the appropriate HTTP or TFTP server. Note: To program the HTTP or TFTP server, access the Network Configuration page. CAUTION: DO NOT remove power from the phone while firmware is downloading or while the phone is rebooting; otherwise, the phone may be severely damaged. To manually upgrade firmware in the phone: 1. On the Web Configuration Tool home page, click Firmware Update. The Firmware Update page appears. 2. In the Automatic Firmware Upgrade section, from the HTTP Automatic Firmware Upgrade list, select Off. 3. From the TFTP Automatic Firmware Upgrade list, select Off. 4. In the Manual Firmware Upgrade section, from the Choice of Download Type list, select the required server type (i.e. HTTP or TFTP). If you select None, then HTTP takes priority over TFTP. 5. Depending on the server type selected in step 4, enter either the URL for the HTTP Server or the URL for the TFTP Server. 6. Do one of the following: (Recommended) To install the latest firmware but leave the existing phone configuration files unchanged, click Update. A confirmation screen appears. To install the latest firmware and reset the existing phone configuration files to new default settings, click Update & Reset Config Files. 7. If you clicked Update & Reset Config Files in step 6, re-enter the SIP authorization password and the PPPoE and user passwords, if required. To re-enter this information, do one of the following: Use the Web Configuration Tool Use the phone's Settings key menu interface (see Configuring in the Main and Boot Programs) Edit a saved configuration file. For more information on configuration files, see About Configuration Files. 8. Click OK. Note: The boot program and boot menu only support IP format for the TFTP server. If a URL is configured, the URL will display as 255.255.255.255 and the boot program will not download from the URL. 5-2 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting Automatically Updating the Firmware To automatically update firmware in the phone: Ensure the HTTP Automatic Firmware Upgrade and TFTP Automatic Firmware Upgrade dropdown lists are set to Auto. Click Apply. Do steps 6 and 7 under Manually Updating the Firmware. Troubleshooting Tip: If your phone displays "SIP MAIN NOT FOUND", it is likely that your system has experienced a power failure. The SIP Phone Boot firmware "borrows" Flash sectors from the SIP Main area during firmware installation. At the end of a normal installation, the sectors are restored without affecting SIP Main. However, if power is removed during Boot installation, then SIP Main is erased and will have to be reinstalled on the phone. 5-3

Mitel SIP Workshop Working with Firewalls on a SIP mode Mitel Phone You can configure the phone to work behind Network Address Translation (NAT) firewalls that are not SIP-aware by enabling the SIP configuration Bypass Firewall NAT feature and configuring the firewall correctly. Mitel SIP phones support two firewall traversal methods: Dynamic firewall traversal Static firewall traversal Dynamic Firewall Traversal The easiest way to configure the SIP phone when running multiple SIP phones behind a NAT firewall is to use dynamic firewall configuration. In order to support this feature, the SIP server must support the following protocols: SIP rport extension SIP outbound extension, or STUN server support Note: The STUN server must be running on the same machine (or in the same subnet) as the SIP server. SIP Phone Configuration 9. On the Advanced Features page, in the Firewall/NAT section, enter the IP address of the STUN server. 10. Click Apply. 11. On the User List Config page, select the appropriate user from the current list. 12. In the Firewall/NAT section, select a SIP Keep Alive Mode option. 13. In the Type field, do one of the following: 14. Select the OUTBOUND/STUN keep-alive mechanism if the server supports SIP outbound extension. 15. Otherwise, select either OPTION, or PING, depending on server capability. Most SIP servers support the OPTION method. 16. In the IP address field, select WAN if you know the phone is surely behind a NAT firewall. Note: In some deployments, the server has its own firewall traversal mechanisms, such as using Session Border Controller (SBC) between the SIP client and SIP server. In this case, the phone may not require the WAN address. If the WAN is selected, then the STUN IP address must be configured on the Advanced Features page. Failure to do so will cause the phone to experience very slow response from the UI. Static Firewall Traversal If the SIP server does not support the STUN or rport SIP extensions, or if you know the public IP address of your NAT firewall, you can configure the SIP phone with the static method. When you use the static method, you need to configure both the SIP phone and your NAT firewall. For this method, you need to refer to your NAT firewall documentation, log in to the phone, and refer to the Web Configuration Tool. 5-4 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting NAT Firewall Documentation 1. Configure the phone to function as a Demilitarized Zone (DMZ) server to the firewall. For instructions about configuring a server, refer to the documentation packaged with your NAT firewall. 2. Proceed to Web Configuration Tool. Web Configuration Tool 1. On the Web Configuration Tool home page, click Network Configuration. 2. On the Network Configuration page, ensure a static IP address, Subnet Mask, gateway and DNS server address are configured. 3. (Optional) From the DHCP drop-down list, select On or Off to enable or disable DHCP. 4. (Optional) From the Address Type drop-down list, select the appropriate DHCP address type. 5. Click Apply, and then click OK in the confirmation window. 6. On the Web Configuration Tool home page, click Advanced Features. 7. On the Advanced Features page, in the Firewall/NAT section, do the following: 8. From the Static NAT Transversal list, select ON. 9. In the WAN IP Address field, enter a valid WAN IP address. 10. Click Apply. Note: If you enable the Static NAT Traversal, it will over write the firewall traversal features you configured in the User Configuration page. Note: This firmware version does not support the TURN or ICE standards so media NAT/Firewall traversal may not work if the phones are behind NAT-unfriendly (symmetric) firewalls. In these cases, the phones depend on the server or on a Session Border Controller (SBC) for NAT/firewall traversal. 5-5

Mitel SIP Workshop SIP Commands BUSY PEER <PEER_NAME> BUSY FORCE PEER <PEER_NAME> RTS PEER <PEER_NAME> LINK STATE PEER <PEER_NAME> LINK STATE ALL LINK STATUS SHOW ACTIVE ALL SHOW ACTIVE PEER <PEER_NAME> SHOW ACTIVE PROFILE <PROFILE_NAME> SNMP COMMON STATS SUMMARY SNMP OTHER COMMON STATS SNMP STATS METHOD <METHOD_NAME> SNMP TIMERS CONFIGURATION SNMP COMMON CONFIGURATION SNMP PORT CONFIGURATION SNMP SUPPORTED METHODS STATS PROFILE <PROFILE_NAME> STATS ALL SIP REGISTRAR CONFIG SIP REGISTRAR STATS SIP REGISTRAR CONTACTS ALL SIP REGISTRAR CONTACTS <USER_NAME> SIP MWI STATS SIP MWI STATS CLEAR SIP MWI SUBSCRIBER INFO ALL SIP MWI SUBSCRIBER INFO <CONTACT NAME> SIP URI TRANS DEBUGON SIP URI TRANS DEBUGOFF 5-6 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting SIP URI TRANS LIST SIP URI TRANS SORTURI SIP UIR TRANS SORTNUM Purpose: Provides maintenance commands for SIP trunks. Details: Command Details Busies the designated peer. This is a courtesy busy out. Existing calls are allowed to complete normally, but new calls are blocked. When a courtesy busy out is used, the number of active calls is displayed. When all calls are completed, the state of the link changes to MAN BUSY. SIP BUSY PEER <PEER_NAME> <PEER_NAME> The peer name is the "Name" from the Network Element form, and the name is case sensitive. Example: sip busy peer SoftSw2A SIP BUSY ( COURTESY ) PEER SoftSw2A succeeded. Link state changes from IDLE to MAN BUSY. SIP BUSY FORCE PEER <PEER_NAME> Forces busy on the designed peer name. The calls in progress are dropped. Returns the busied peer to service. Example: SIP RTS PEER <PEER_NAME> sip rts peer SoftSw2A SIP RTS PEER SoftSw2A succeeded. Link state changes from MAN BUSY to IDLE. Displays the link state for the selected peer name. SIP LINK STATE PEER <PEER_NAME> The state can be one of: "IDLE" - Initial state; no calls have been made on the link. "IN SERVICE" - The link is in service. 5-7

Mitel SIP Workshop "OUT OF SERVICE" - A temporary or permanent network outage/error was detected on the link. "VALIDATING" - A new call is being attempted and is bringing the link from OUT OF SERVICE to IN SERVICE, if possible. "MAN BUSY PENDING" - The link is being busied out, and no new calls can be made. One or more existing calls are still up. "MAN BUSY" - The link is busied. "UNKNOWN" - Unexpected error on the system. Displays the link state for all peers. Example: SIP LINK STATE ALL sip link state all There are 2 SIP link(s) programmed. Peer 41 state is IN SERVICE. Peer SoftSw2A state is IDLE. SIP LINK STATUS SIP SHOW ACTIVE ALL SIP SHOW ACTIVE PEER <PEER_NAME> SIP SHOW ACTIVE PROFILE <PROFILE_NAME> SIP SNMP COMMON STATS SUMMARY Displays the number of established links and nonestablished links. Displays a summary of all the active calls. Displays an active call for the selected peer name. Displays active calls for a specific profile name. Displays a summary of common statistics. SIP SNMP OTHER COMMON STATS Displays other common statistics. Displays the total number of incoming and outgoing requests for a particular method. SIP SNMP STATS METHOD <METHOD_NAME> Method names: INVITE, ACK, CANCEL, BYE, INFO, UPDATE, PRACK, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, REFER SIP SNMP TIMERS CONFIGURATION Displays the values for the configured timers. SIP SNMP COMMON CONFIGURATION Displays values for the common configuration. 5-8 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting SIP SNMP PORT CONFIGURATION SIP SNMP SUPPORTED METHODS SIP STATS PROFILE <PROFILE_NAME> SIP STATS ALL Displays the list of ports and transport types being used by the SIP module. Displays the list of all supported SIP methods. Displays statistics data for a specific profile name, including the number of active calls, rejected calls, incoming and outgoing non-authorized calls, incoming and outgoing authorized calls, incoming and outgoing authorized failures. Displays statistics data for all SIP links in the system, including the number of active calls, rejected calls, incoming and outgoing non-authorized calls, incoming and outgoing authorized calls, incoming and outgoing authorized failures. Displays configuration data for the Registrar: Domain name- the user name and IP address. Authentication - whether Authentication is enabled. Third Party Registrations - whether third-party registrations are allowed. Default Expiry - the default time interval for registrations to be active. Minimum Configurable Expiry Duration - the minimum expiry duration that may be requested by a User Agent for a particular contact. SIP REGISTRAR CONFIG Maximum Configurable Expiry Duration - the maximum expiry duration. Registrar Configuration ----------------------- Domain: mycompany.com IP Address 10.37.105.20 Authentication Enabled: yes Allow Third Party Registrations: yes Default Expiry Duration: 3600 seconds Minimum Configurable Expiry Duration: 300 seconds Maximum Configurable Expiry Duration: 7200 seconds SIP REGISTRAR STATS Displays the following statistics data: 5-9

Mitel SIP Workshop Current - the current number registered. Accepted - the number of registrations that have been accepted. Refreshed - the number of times registrations have been refreshed. Total failures - the total number of registration failures. Rejected registrations - number of rejected registrations caused by some system or messaging error, and not specifically by one of the other categories. Failure responses - the failure response with the associated SIP failure message (e.g., 400 bad request is a SIP error message). Registrar Statistics -------------------- Current: 3 Accepted: 3 Refreshed: 2 Total Failed: 0 (Rejected 0) (400 Bad Request 0) (403 Forbidden 0) (404 Not Found 0) (423 Min Expires 0) Displays the following for all users: The first number - local identifier followed by an address in brackets. SIP REGISTRAR CONTACTS ALL The state - either registered or pending. The address - each contact and the address that was registered (the examples only list a single contact). Time left - the time before expiry. Expires - the expiry time is the total time negotiated. Priority - the contact's priority. 5-10 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting Port - the registered port. Registrar Entry(s) -------------------- 3 (0xa3cda40) State:Registered Addr:sip:40500@10.37.105.20 Contact1: sip:40500@10.37.102.44 Time Left:287 Expires:400 Priority:0 Port:5060 2 (0xa3be408) State:Registered Addr:sip:40501@10.37.105.20 Contact1: sip:40501@10.37.102.43 Time Left:280 Expires:400 Priority:0 Port:5060 1 (0xa3be3e0) State:Registered Addr:sip:40503@10.37.105.20 Contact1: sip:40503@10.37.102.42:5060 Time Left:3208 Expires:3600 Priority:0 Port:5060 Displays the same information as SIP REGISTRAR CONTACTS ALL, except displays only those entries that have the user_name in the Addr: For example, if 40500 is the contact name, the following is displayed: SIP REGISTRAR CONTACTS <USER_NAME> Registrar Entry(s) -------------------- 3 (0xa3cda40) State:Registered Addr:sip:40500@10.37.105.20 Contact1: sip:40500@10.37.102.44 Time Left:201 Expires:400 Priority:0 Port:506 Scheme:1 Displays the following for all users: First number - local identifier followed by an address in brackets. SIP REGISTRAR CONTACTS ALL State - either registered or pending. Address - the address that was registered and each contact registered is printed. (All these examples only list a single contact.) 5-11

Mitel SIP Workshop Time left - the time before expiry. Expires - the expiry time is the total time negotiated. Priority - the contact's priority. Port - the registered port. Registrar Entry(s) -------------------- 3 (0xa3cda40) State:Registered Addr:sip:40500@10.37.105.20 Contact1: sip:40500@10.37.102.44 Time Left:287 Expires:400 Priority:0 Port:5060 2 (0xa3be408) State:Registered Addr:sip:40501@10.37.105.20 Contact1: sip:40501@10.37.102.43 Time Left:280 Expires:400 Priority:0 Port:5060 1 (0xa3be3e0) State:Registered Addr:sip:40503@10.37.105.20 Contact1: sip:40503@10.37.102.42:5060 Time Left:3208 Expires:3600 Priority:0 Port:5060 Displays the following information: Current - the number of current MWI subscriptions. In the following example, three phones are registered, and two are also subscribed for MWI. Successful - the number of successful subscriptions. SIP MWI STATS Failed - the number of failures. MWI Subscription Statistics -------------- Current: 2 (out of 3 Registrations) Successful: 6 Failed: 2 SIP MWI STATS CLEAR Clears the number of successful and failed registrations. The following message is displayed: "MWI Subscription Statistics Cleared" 5-12 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting Displays the following information about the current state of MWI for all endpoints: State - MWI state of the subscribers. SIP URIs - SIP URIs for the endpoints. Time left - amount of time (in seconds) left in the subscriptions. Expires - the currently negotiated expiry time for this phone. SIP MWI SUBSCRIBER INFO ALL In the example below, the first entry shows the MWI lamp is OFF, and in the second entry, it's ON. The phone's address is printed along with the time left before the subscription expires. MWI Entry(s) ------------ OFF sip:40500@10.37.105.20 Time Left:96 Expires:400 ON sip:40501@10.37.105.20 Time Left:288 Expires:400 Displays the same information as SIP MWI SUBSCRIBER INFO ALL for a particular user, except the command uses a string to search for entries that match. For example, using 40500 as the contact name, it displays only the first entry. Using 405 displays both entries below. SIP MWI SUBSCRIBER INFO <USER_NAME> MWI Entry(s) ------------ OFF sip:40500@10.37.105.20 Time Left:96 Expires:400 ON sip:40501@10.37.105.20 Time Left:288 Expires:400 SIP URI TRANS DEBUGON SIP URI TRANS DEBUGOFF SIP URI TRANS LIST Turns on debug tracing to display URI/Number translations on a per call basis. Turns off debug. Displays an unsorted list of all URI/Number entries in the translation table. 5-13

Mitel SIP Workshop SIP URI TRANS SORTURI SIP URI TRANS SORTNUM Displays a sorted list (by URI) of all URI/Number entries in the translation table. Displays a sorted list (by number) of all URI/Number entries in the translation table. 5-14 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting Lab 1 Upgrading the Mitel Firmware In this lab you will: Download the latest firmware from Mitel OnLine Upgrade a Mitel phone to the latest firmware Download latest firmware from Mitel OnLine Step Task Expected Result/Observation 1 Login to to Mitel with your MOL Account 2 Select Support > Software Downloads 3 Click the 3300 ICP category Click SIP Phone Software Downloads - SD 4 Select the most recent SIP Firmware Zip file and save it to C:\Mitel Academy\SIP Workshop 5 Extract the content of the ZIP file 6 Using Internet Explorer, open ftp://192.168.y.2, where y is you subnet id Open the TFTP Folder Copy the extracted SIP Firmware files to the 3300 7 You are now ready to configure the phone Upgrade the Firmware on a Mitel 5224 Step Task Expected Result/Observation 1 Reset the phone and hold down the volume UP button (^) 2 When the menu appears scroll through and select Network Parameters > modify static values 5-15

Mitel SIP Workshop 3 Scroll through the list filling in the following details: TFTP Server: 192.168.y.2, where y is your subnet id TFTP Port: 69 Exit the menu and Save the setting Do not reboot at this stage 4 Continue to scroll through the menu until Upgrade setting Change TFTP upgrade to yes Save the settings and reboot the phone 5 Choose Yes to accept he firmware upgrade The phone will now be upgraded. This may take a few minutes 6 The phone now has the latest version of SIP firmware installed 5-16 Module 5 Maintaining and Troubleshooting

Maintaining and Troubleshooting Lab 2 Creating a trace using Wireshark 5-17

Mitel SIP Workshop Lab 3 Analyzing SIP Traffic 5-18 Module 5 Maintaining and Troubleshooting

Appendix 6

Mitel SIP Workshop Appendix

Detailed SIP Forms A Objectives When you finish this module, you will be able to: Describe the purpose of each of the settings for the SIP Device Capabilities Assignment form.

Mitel SIP Workshop A - 2 Appendix A Detailed SIP Forms

Detailed SIP Forms SIP Device Capabilities Assignment Purpose This form provides configuration options that can be applied to various types of SIP devices. The association between the SIP device and the form is similar to how the Class of Service options work. The SIP Device Capabilities number provides a SIP profile that can be applied to particular SIP devices to allow for alternate capabilities as recommended through the Mitel interop process. A SIP phone can only be associated with a single SIP Device Capabilities Assignment form, though a form may be assigned to several SIP phones, for example, one SIP Device Capabilities Assignment form can be assigned to all of one type of generic SIP phone. The SIP Device Capabilities forms are shared across all nodes in a cluster by default (see SDS sharing in the MCD Help for more information). A- 3

Mitel SIP Workshop Details System-generated, protected field. Indicates the number of the SIP Device Capabilities that you want to edit. There can be up to 64 different device capability numbers. This number is similar to a Class of Service Option. All entries contain system defaults, unless otherwise modified. Enter a meaningful name to describe the type of SIP devices assigned to this number. The comment can be up to 15 characters in length. Call Routing and Administration Options Select the network element that is being used as the Outbound Proxy Server for SIP devices. The Outbound Proxy Server will then be added to the Trusted Domains for Registration. The default is to leave this field blank, which means that no Outbound Proxy Server is deployed between SIP devices and the 3300 ICP. Note If no Outbound Proxies are defined in the Network Elements forms, then there will be no option to select an Outbound Proxy. Some SIP phones support the Transfer and Conference features independently of the 3300 ICP system. Select "Yes" to use the Transfer and Conference feature functionality that is provided by the SIP phone. Select "No" to use the Transfer and Conference features that are provided by the 3300 ICP. SDP Options When enabled, the 3300 will allow the device to negotiate more than one active m-line for the Session Description Protocol. For devices capable of negotiating multiple m-lines, such as audio, image, video, and applications, the recommended setting is Yes. Select Yes to always insert SDP into the initial Invite message. This option allows inter-working with SIP peers that require SDP in the Initial Invite. In many situations, the 3300 ICP already includes SDP in the initial invite. The forced SDP may contain a default inactive SDP if the SDP is not available at call setup time. Select Yes to prevent sending SDP renegotiation messages to the peer device when the peer is on hold. A - 4 Appendix A Detailed SIP Forms

Detailed SIP Forms Select Yes to prevent the IP Address of 0.0.0.0 from being used as the connection address in the SENDONLY or INACTIVE SDP sent by the 3300 ICP. Enable this option if there are specific 0.0.0.0 issues detected with the SDP. Enable this option to force the use of the same codec--for example, G.729--in both incoming and outgoing directions. When a SDP negotiation is answered with a list of codecs it can be unclear which codec will be used in either direction. This option will send a Re-invite with a single codec if the 3300 ICP detects this situation to ensure there is no misunderstanding. Select "Yes" to treat identical SDP offers received in the same session as a session refresh instead of responding with a new answer that repeats the previous SDP. By default, all SDP Offers (duplicate or not) are treated as a new offer, resulting in audio renegotiation and the restart of RTP streaming. This option should be enabled if changing/restarting RTP streaming causes audio issues when the remote peer simply refreshes the media connection. Select Yes to allow the 3300 ICP to minimize the use of INACTIVE messaging if it is not fully supported by the service provider. While media connections are transitioning (hold/transfer/conference, etc.), or in some cases at call setup, the media may be temporarily unavailable causing the 3300 ICP to send an inactive SDP. If this option is set to Yes the 3300 ICP will instead attempt to use the IP 0.0.0.0 or mark streams as sendonly/recvonly to avoid the use of inactive which is not supported by some SIP Peers. Signaling and Header Manipulation This field sets the minimum time period in seconds allowed between Register requests to the primary ICP and secondary ICP. The range is from 120 to 3,600 seconds. CAUTION: Frequent Register requests from large numbers of SIP devices can significantly degrade system performance. Therefore, we recommend that the Minimum Registration Period be set to no less than the default of 300 seconds. Note that in cases where there are few SIP devices, or where most SIP devices support the P-Alternate-Server header, you can set the minimum registration period to a lower value. Set the SIP timeout value (90 to 9999 seconds). If the device does not respond within the allocated time, the session (call) is torn down. Set the Session Timer to 0 to disable it. It is recommended that this be set to a non-zero value unless there is a specific reason to disable it. The benefit of this option is that it can help clear calls that get stuck due to signaling errors. If the peer responds with a 422 Session Interval too Small, you may want to increase the session timer to the value indicated by the peer to minimize delays in call setup. Select Yes to disable the use of reliable provisional responses (PRACK) on outgoing and incoming calls, unless the required header is used on incoming calls. A- 5

Mitel SIP Workshop Network Element Sip Peer Purpose The Network Element Assignment form manages all elements within a network. An element can be a Mitel 3300 ICP, an SX-2000, a SIP Peer or Outbound Proxy It is used to view, define, and manage the basic attributes of network elements, such as name, IP address, type, data-sharing status, and zones. When working with SIP Trunks, it is used to provide details of the local switch and allows for the creation of a network element for each SIP peer, gateway or Service provider. Outbound Proxies can also be created, if one exists in the network. Network Elements can be identified using IP Addresses or Fully Qualified Domain Names (FQDN). DNS must be configured correctly to allow FQDN s to be resolve to IP Addresses. A - 6 Appendix A Detailed SIP Forms

Detailed SIP Forms Details Enter a unique name of up to nine characters for the network element. Start the name with an upper case letter; enter all other letters in lower case. Digits are allowed but not spaces or nonalpha characters. This naming convention must be followed if Enterprise Manager will be used to manage the network element. This name displays in read-only format in the system name field in the local element's System Options Assignment form. Defines the type of network element (for example, 3300 ICP). Select the network element type from the drop-down list. The type selected determines the specific information that displayed to be filled in. For a SIP peer, make sure you select Other, and the choose the option for SIP Peer. Defines the Fully Qualified Domain Name (FQDN) or IP Address of the Network Element. This field is disabled for the local network element. Typically, you should enter an IP Address. However, if the system will support SIP devices and if the SIP Service Provider uses the Fully Qualified Domain Name (FQDN) instead of an IP Address to communicate with the system, enter the FQDN. Displays "True" if the selected network element is local. Displays "False" if the selected network element is not the local element. This field should only display "True" for the element you are logged in to. If the local element is sharing data with a remote element, this field displays the remote element's software version. The version number is cleared if you disable SDS on either element, or disable data sharing between the elements. The zone specified for the IP voice media stream originating or terminating at the selected ICP. It defaults to 1. The zone range is 1-250. Click the check box to make the network element a SIP Peer and then fill in the related SIP Peer Specific fields that display. SIP Peer Specific The list of transport parameters used in the SIP URI: UDP/TCP/TLS. Use TLS if you require encryption between 3300 ICP devices in the same network. TLS is not supported across firewalls. A- 7

Mitel SIP Workshop Enter the port number of the SIP Peer. Enter the IP address or the Fully Qualified Domain Name (FQDN) of the SIP proxy server of the peer network. Do not fill in this field if an External SIP Proxy is not in use. The list of transport parameters used in the SIP URI: UDP/TCP/TLS. Enter the port number of the proxy. Enter the IP address or the Fully Qualified Domain Name (FQDN) of the SIP registrar of the peer network. Do not fill in this field if an External SIP Proxy is not in use. The list of transport parameters used in the SIP URI: UDP/TCP/TLS. Enter the port number of the SIP Peer. Defines SIP peer status, which can be Auto-Detect/Normal, Disabled, or Always Active. Auto-Detect/Normal periodically sends OPTIONS messages to test to see if the link to the remote SIP Peer is alive. When the link is detected to have gone away the SIP Module marks the link out-of-service and periodically sends OPTIONS messages to restore service. Setting the Always Active option causes the SIP Module to assume that the link is available unless you choose another option. Use the Disabled option to temporarily disable a trunk connection A - 8 Appendix A Detailed SIP Forms

Detailed SIP Forms Network Element Outbound Proxy Purpose A Proxy Server is an intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced either internally or by passing them on, possibly after translation, to other servers. A Proxy interprets, and, if necessary, rewrites a request message before forwarding it. Details Enter a unique name of up to nine characters for the network element. Start the name with an upper case letter; enter all other letters in lower case. Digits are allowed but not spaces or nonalpha characters. This naming convention must be followed if Enterprise Manager will be used to manage the network element. This name displays in read-only format in the system name field in the local element's System Options Assignment form. Defines the type of network element (for example, 3300 ICP). Select the network element type from the drop-down list. The type selected determines the specific information that displayed to be filled in. A- 9

Mitel SIP Workshop Defines the Fully Qualified Domain Name (FQDN) or IP Address of the Network Element. This field is disabled for the local network element. Typically, you should enter an IP Address. However, if the system will support SIP devices and if the SIP Service Provider uses the Fully Qualified Domain Name (FQDN) instead of an IP Address to communicate with the system, enter the FQDN. Displays "True" if the selected network element is local. Displays "False" if the selected network element is not the local element. This field should only display "True" for the element you are logged in to. If the local element is sharing data with a remote element, this field displays the remote element's software version. The version number is cleared if you disable SDS on either element, or disable data sharing between the elements. The zone specified for the IP voice media stream originating or terminating at the selected ICP. It defaults to 1. The zone range is 1-250. Outbound Proxy Specific The list of transport parameters used in the SIP URI: UDP/TCP/TLS. Enter the port number of the Service Provider or the gateway. A - 10 Appendix A Detailed SIP Forms

Detailed SIP Forms SIP Peer Profile A- 11

Mitel SIP Workshop Enter an alphanumeric string up to nine characters for the SIP Peer Profile. Select the appropriate Network Element name (programmed in the Network Element form) from the pull-down list. Local Account Information Enter the DIDs assigned by the Service Provider if registration is required. The default is blank. Enter one or more user names. The field accepts a maximum of 60 characters. The maximum number of alphanumeric characters allowed per user name is 26. If using a range, you must use only digits 0 through 9 with a dash (-) separating the first number in the range and the last. You can enter a mix of numerical ranges and single usernames (for example, "6135554000-6135554400, 6135554500"). Use a comma to separate user names and ranges. The first and last characters cannot be a comma or a dash. A - 12 Appendix A Detailed SIP Forms

Detailed SIP Forms For Service Providers and SIP Services that require the 3300 to register, this field is used to indicate the user names that they wish us to register with. Normally (when required) this is a single user name but it could also refer to a range of DIDs being supplied by the service provider. Select the address type for the local host. Two types are available: FQDN: the Fully Qualified Domain Name (FQDN) of the local host, and the DNS name from the System IP Configuration form. IP: from the System IP Configuration form. Note: Ensure that the 3300 ICP and the Service Provider are using the same Address Type in their SIP messages. If the 3300 ICP uses FQDN, then the Service Provider must use FQDN. Call Routing and Administration Options Enter the Interconnect Restriction number that is used to restrict device interconnections. Enter the maximum allowable number of incoming and outgoing simultaneous calls for this peer. Note: Set the maximum number of calls per link the same on both ends of the link. Select the network element to be used for the Outbound Proxy Server. Routing Prefix For Unknown URIs Enter a Routing Prefix to be used to route redirected or referred calls back through ARS Automatic Route Selection (ARS) simplifies local and long distance dialing by automatically selecting the most convenient and cost-effective route, and by inserting and/or deleting digits for proper routing. An alternative name is Call Route Selection. and then out a SIP Trunk on this local 3300 ICP. If this option is enabled, the 3300 ICP checks the Contact header in the 300, 301 or 302 messages for an unknown URI A short character string used to identify a name or resource on the Internet. or checks the Refer-To header in a REFER message. If an unknown URI is detected, the 3300 ICP then maps the unknown URI to a dynamic DN which has the programmed prefix. When the call is re-routed to the SIP trunk, the dynamic DN is mapped back to the Unknown URI in the Invite message. The default is to leave this field blank and to disallow routing to unknown URIs. A- 13

Mitel SIP Workshop Use the same routing prefix added here in the ARS Digits Dialed Assignment form. For example, if you specify the prefix for unknown URIs as "7777" here, you must set the same prefix in the ARS Digits Dialed Assignment form. Enter the SMDR tag number. The range is 1-9998. This tag number is used in SMDR logs for both incoming and outgoing calls when SMDR is enabled. Enter the Trunk Service number where the COS/COR and incoming digit modification setting are set. Use this field to group together devices to which compression policies can be applied. Select Yes to allow the administrator to specify the alternate domain to be used for this peer. Enter the alternate domain name. For example, when using the Microsoft Live Communications Server (LCS), the Office Communicator endpoint on the peer side may be in a different domain than the LCS. When enabled, this option will insert the alternate FQDN or IP Address into the To header. In addition, the FQDN or IP Address may be used to find the appropriate Peer Profile on incoming calls. Enter the Fully Qualified Domain Name or IP Address of the Alternate Destination Domain. Select Yes to enable special handling of re-invite collisions. Normally, when a re-invite collision is detected, both re-invite messages are rejected with a 491. With this option enabled, the incoming re-invite wins. If enabled, calls received on this trunk will be considered private or non-public. The purpose of this option is to allow you to set SIP trunks to private for a small SIP gateway. Enabling this option allows the system to handle CPN substitution properly at ISDN and SIP interfaces. When trunks are connecting small gateways or devices that have non-public numbers, this option should be set to Yes (Private/non-Public). When set to Yes, calls delivered to call control will be treated as non-public trunk. If the call is directed to an ISDN type interface, the calling party number from a public trunk may pass 'as is' out to the network. The private/non-public number may undergo CPN substitution before being sent out to the network. Enable this option to route all incoming calls to the called individual user based on the information present in the To URI instead of the Request URI. In most cases, the information in these two lines is the same. When the two lines differ, this option is used to select which one should be used. A - 14 Appendix A Detailed SIP Forms

Detailed SIP Forms Calling Line ID Options Enter the default CPN up to 26 alphanumeric characters (a-z, A-Z and 0-9). This alphanumeric string is used in outgoing calls to replace the calling party number and on incoming calls to replace the called party number when a match is not found in the URI/Number Translation Form or the Outgoing DID ranges on the SIP Peer Profile Form. Normally on outgoing calls (unless restricted or private) the Default CPN would appear in the From header if a match is not found elsewhere. If the "Use P-Asserted-Identity Header" option is set, the P-Asserted-Identity will be included and it will contain the Default CPN if a match is not found elsewhere. A "Privacy: id" header will also be included if the CPN is restricted or the number is marked private. If the "Use P-Preferred-Identity Header" option is set and the "Use P-Asserted-Identity Header" is not set, the P-Preferred-Identity will be included and will contain the Default CPN. For incoming calls, the ringing or answering party's number is replaced with the default CPN when a match is not found in the URI/Number Translation Form or the Outgoing DID ranges. This number will only be included in the P-Asserted-Identity header if the option is enabled. To substitute specific numbers to DIDs you may add numbers into the DID ranges for CPN Substitution form, and then select the appropriate numbers in the Outgoing DID Ranges in this form (SIP Peer Profile). Note: Public numbers may be substituted by the Default CPN unless the option "Public Calling Party Number Passthrough" is set. Select to force anonymous@anonymous.invalid in the From header on outgoing calls and to prevent CPN substitution on incoming calls. If the "Use P-Asserted-Identity Header" option is set the P-Asserted-Identity will be included and it will contain the calling party number or some substitution. A "Privacy: id" header will also be included in this case to indicate that the identity should be kept private. If an incoming call is received by the 3300 ICP through an ISDN trunk or a Public SIP trunk, you can allow the public number to be passed through the 3300 ICP when it leaves via a SIP trunk. Enable this option to allow the public CPN to be passed through the 3300 ICP and not substituted with the default CPN (normal behaviour). Note: Passing public numbers through the 3300 ICP is restricted in some areas. Enable this option to use the diverting/forwarding party as the CPN on the outbound SIP call instead of the original calling party number. The party at the final call destination sees the call as being from the diverting party because the original calling party information is not provided. A- 15

Mitel SIP Workshop Authentication Options Enter the authentication user name (up to 48 characters long) used to authenticate incoming/outgoing calls. The user name is also used to authenticate the registration, if applicable. Enter the authentication password associated with the Authentication User Name (hidden). The password is also used to authenticate the registration, if applicable. Re-enter the authentication password to confirm it. Select the type of authentication challenges for incoming calls: No Authentication: No authentication performed. Challenge-Based Authentication: All incoming calls are challenged with digest authentication utilizing the username and password programmed for this profile. Validate Address: The IP address or FQDN of all incoming calls is validated against the entry programmed in the Network Element form. SDP Options Enable this option for the 3300 ICP to allow the peer to negotiate more than one active m-line (media description line). For peers capable of negotiating multiple m-lines, such as audio, image, video, and applications, the recommended setting is enable or Yes. Select No to disable sending Mitel Proprietary lines within the SDP. Mitel proprietary SDP information is only of value when SDP from one 3300 ICP is delivered to another 3300 ICP. It is usually acceptable to select No when inter-working with 3rd party equipment. The other benefit of selecting No is that it reduces the SDP size. Select Yes to always insert SDP into the initial Invite message. This option allows inter-working with SIP peers that require SDP in the Initial Invite. In many situations, the 3300 ICP already includes SDP in the initial invite. The forced SDP may contain a default inactive SDP if the SDP is not available at call setup time. Enable this option if the Initial Outgoing Invite must contain a "sendrecv" Session Description Protocol (SDP) message that identifies the IP address/port of the calling device. This option takes precedence over the Force sending SDP in initial Invite message option (which may send an inactive SDP or an IP 0.0.0.0 SDP). The 3300 provides the connection information on the outgoing call by receiving a Fake Answer prior to sending the Initial Invite. Select to send UDP packets every 30 seconds to the Peer s Audio IP Address and port to keep a pinhole open on a NAT firewall. The packets are sent whether the connection is one-way or two-way. A - 16 Appendix A Detailed SIP Forms

Detailed SIP Forms If enabled, a SENDONLY or INACTIVE sdp that is sent by the 3300 ICP will not contain the IP Address 0.0.0.0 as its connection address. If available, it will use the endpoint's IP address; otherwise, it will send the 3300's IP address + port 9000. Enable this option if there are specific 0.0.0.0 issues detected with the SDP. Enable this option to force the use of the same codec--for example, G.729--in both incoming and outgoing directions. When a SDP negotiation is answered with a list of codecs it can be unclear which codec will be used in either direction. This option will send a Re-invite with a single codec if the 3300 ICP detects this situation to ensure there is no misunderstanding. Select "Yes" to treat identical SDP offers received in the same session as a session refresh instead of responding with a new answer that repeats the previous SDP. By default, all SDP Offers (duplicate or not) are treated as a new offer, resulting in audio renegotiation and the restart of RTP streaming. This option should be enabled if changing/restarting RTP streaming causes audio issues when the remote peer simply refreshes the media connection. Enable this option if the Service Provider to which this peer is connected requires a packet rate other than the standard 20 ms rate in both the transmit and receive media streams. The following Mitel devices and applications will support variable packetization rates: E2T Ethernet to TDM converter. Gateway between the traditional circuit-switched, time division multiplexed telephone systems and IP devices on the Ethernet. 5304 this set will support the full range of packetization rates on its first release 5215 5220 5212/5224 5312/5324 scheduled for release in the R9.0 project 5235 5330/5340 5550 IP Console Teleworker Unified Communicator Mobile Note: Disabling 'RTP Packetization Rate Override' allows the system to operate at ANY packetization rate. It does not force it to operate at the greyed-out rate in the RTP Packetization Rate list box. If the 'Force RTP Packetization Rate" option has been set to 'Yes", the SIP Module will force all calls involving this trunk to use the packetization rate specified in the 'Packetization Rate" option. The calls will be forced to use the specified rate in both the transmit and receive streams. A- 17

Mitel SIP Workshop Rejected calls or audio problems resulting from an unsuccessful packet rate negotiation will generate Media Negotiation maintenance logs. For more information, see Maintaining the SIP Interface Special Handling of Offers in @xxx response (invite) Select Yes to allow the 3300 ICP to minimize the use of INACTIVE messaging if it is not fully supported by the service provider. While media connections are transitioning (hold/transfer/conference, etc.), or in some cases at call setup, the media may be temporarily unavailable causing the 3300 ICP to send an inactive SDP. If this option is set to Yes the 3300 ICP will instead attempt to use the IP 0.0.0.0 or mark streams as sendonly/recvonly to avoid the use of inactive which is not supported by some SIP Peers. Signaling and Header Manipulation Options Set the SIP session timeout value (in seconds). If the peer does not respond within the allocated time, the session (call) will be torn down. The default is 90. The range is from 90-9999. Setting the Session Timer to 0 disables session timeout. It is recommended that this be set to a non-zero value unless there is a specific reason to disable it. The benefit of this option is that it can help clear calls that get stuck due to signaling errors. If the peer responds with a 422 Session Interval too Small, you may want to increase the session timer to the value indicated by the peer to minimize delays in call setup. Enable this option to construct the contact address in 180 and 200 messages using the Request URI Address received in the initial invite. This option should only be enabled in those situations where the Contact needs to be based on the address received in the Request URI. Select Yes to disable the use of reliable provisional responses (PRACK) on outgoing and incoming calls, unless the Required Header is received on incoming calls. Most Peers now support PRACK and this can be useful in interoperability scenarios with the PSTN (see RFC 3262). If the SIP Peer also supports PRACK, it is recommended that this option be set to No. Select Yes to enable adding '+' to the Called and Calling Party Numbers generated by the 3300 ICP. Select Yes for the 3300 ICP to ignore the loose routing indicator and use strict routing instead. Enabling this option is not recommended unless required by the SIP Peer. See RFC 3261 for more information on Loose Routing. Select Yes to enable sending the P-Asserted Identity header within SIP messages. This option is used to convey identity information both at call setup and during a call. When privacy is enabled, or CPN is restricted, this field will still contain calling/called party information. The 3300 ICP relies on the use of the Privacy: id header to inform other SIP Peers that this information is to be kept private. If the peer is not trusted, select No for this option. A - 18 Appendix A Detailed SIP Forms

Detailed SIP Forms If you enable this option, the system uses the Default CPN data to build the P-Preferred-Identity header. No display name is shown in this header. A P-Preferred-Identity will not be included in messaging if the Use P-Asserted-Identity option is also enabled. One purpose of this header is to provide a company or link/peer number. For example, the From header may include the DID of the person making the call but the P-Perferred-Identity header may contain the main company DID. Select Yes to use the restricted character set "0123456789abcdef" for creating nonce and cnonce strings used in authentication. This setting is used for SIP Servers with limited ascii character support and reduced authentication security. When this option is enabled (set to Yes) the Address used in the SIP From header line will no longer be the physical 3300 IP address or FQDN. Instead the address will be replaced by the address to which the outgoing call is sent. Some providers require this for authentication purposes as it makes it look like the 3300 ICP is in the same domain as the SIP Server or SBC (Session Border Controller). A- 19

Wireshark B Objectives When you finish this module, you will be able to: Install Wireshark Configure Wireshark for a capture Read a capture

Mitel SIP Workshop B - 2 Appendix B Wireshark

Wireshark Overview Note To get the latest version of Wireshark, visit http://www.wireshark.org/download.html B - 3

Mitel SIP Workshop Installing Wireshark Installing Wireshark 1.0.2 Double click on the Installer file, to start the installation wizard. Click Next > Read the License and Click I Agree when complete B - 4 Appendix B Wireshark

Wireshark Select the components you would like to install, the default choice works well. Click Next > Select which shortcuts you would like to have, Click Next > Choose the location for the program files, Click Next > B - 5

Mitel SIP Workshop Choose whether to install WinPcap or not and whether to make it run as a service. WinPcap is needed, but not necessary as a service if your account has administrative permissions on the local machine. Click Install. Installation will begin. After a few seconds WinPcap will begin a new wizard. B - 6 Appendix B Wireshark

Wireshark Click Next > to begin the WinPcap installation wizard Click Next > B - 7

Mitel SIP Workshop Read the License agreement and the click I Agree WinPcap will now be installed. B - 8 Appendix B Wireshark

Wireshark Click Finish Wireshark will have completed its install in the background. Click Next > B - 9

Mitel SIP Workshop Check the Run Wireshark 1.0.2 box to start Wireshark. Click Finish Wireshark 1.0.2 is now installed. Wireshark can be started from the Start menu B - 10 Appendix B Wireshark

Wireshark B - 11

Mitel SIP Workshop Setting up a Trace in Wireshark Creating a capture using wireshark 1.0.2 Open Wireshark Select the Capture menu > Interfaces For a quick start, Click the Start button. Wireshark will begin to capture and display packets B - 12 Appendix B Wireshark

Wireshark To view a packet, simply select it. There are three windows to view Packet List, Packet Details and Packet bytes. By selecting a packet form the top window (Packet List) the details are displayed in the middle (Packet details) and bottom (Packet bytes) windows. Select the Stop the running live capture button. B - 13

Mitel SIP Workshop You have now completed a trace. To save the file click File > Save B - 14 Appendix B Wireshark

Wireshark B - 15

Mitel SIP Workshop Viewing SIP Traces with Wireshark Step 1: Use filter option In the Filter field below, entering "sip" (as shown) allows Ethereal to display all SIP related traces. Step 2: Click Statistics to identify the number of SIP call setup in the ethereal trace If you select "Statistics"-->VOIP calls, the Ethereal application will provide a summary of all VOIP calls Step 2a: Once you have selected VoIP calls, Ethereal will go through the capture and summarize as follows: (as an example) In this example, Ethereal has detected 2 VOIP calls using the SIP protocol. Note: the numbers have been manually modified and are intended for demo only. B - 16 Appendix B Wireshark

Wireshark Step 3: Graphical analysis If you highlight one of VOIP calls above, and click the "Graph" button, you see a graphical SIP output as follows: At this point, you can analyze based on symptoms and SIP protocol. B - 17