MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 4.1 for use with SKYPE SIP Trunking. SIP CoE

Transcription

1 MITEL SIP CoE Technical Configuration Notes Configure MCD 4.1 for use with SKYPE SIP Trunking SIP CoE

2 NOTICE The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks Corporation (MITEL ). The information is subject to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate such changes. No part of this document can be reproduced or transmitted in any form or by any means - electronic or mechanical - for any purpose without written permission from Mitel Networks Corporation. TRADEMARKS Mitel is a trademark of Mitel Networks Corporation. Windows and Microsoft are trademarks of Microsoft Corporation. Other product names mentioned in this document may be trademarks of their respective companies and are hereby acknowledged. Mitel Technical Configuration Notes Configure MCD 4.1 for use with SKYPE SIP Trunking April , Trademark of Mitel Networks Corporation Copyright 2010, Mitel Networks Corporation All rights reserved ii

3 Table of Contents OVERVIEW... 1 Interop History...1 Interop Status...1 Software & Hardware Setup...1 Tested Features...2 Device Limitations and Known Issues...3 Network Topology...4 CONFIGURATION NOTES MCD Configuration Notes...5 Network Requirements... 5 Assumptions for the 3300 MCD Programming... 5 Licensing and Option Selection SIP Licensing... 6 Class of Service Assignment... 7 Network Element Assignment... 9 Network Element Assignment (Proxy) Trunk Service Assignment SIP Peer Profile Digit Modification Number Route Assignment ARS Digits Dialed Assignment T.38 Fax Configuration Zone Assignment iii

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5 Overview This document provides a reference to Mitel Authorized Solutions providers for configuring the Mitel 3300 MCD to connect to SKYPE. The different devices can be configured in various configurations depending on your VoIP solution. This document covers a basic setup with required option setup. Interop History Version Date Reason 1 Mar. 23, 2010 Initial Interop with Mitel _1 and SKYPE SIP trunk Interop Status The Interop of SKYPE trunk has been given a Certification status. This service provider or trunking device will be included in the SIP CoE Reference Guide. The status SKYPE trunk line achieved is: The most common certification which means SKYPE SIP trunk 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. Software & Hardware Setup This was the test setup to generate a basic SIP call between SKYPE trunk line and the 3300 MCD. Manufacturer Variant Software Version Mitel 3300 MCD Mxe Platform _1 Mitel Minet sets: 5340, 5215, 5330 Mitel MBG - Teleworker Mitel Mobile Extension Mitel NuPoint NuPoint voice mail server V

6 Tested Features This is an overview of the features tested during the Interop test cycle and not a detailed view of the test cases. Please see the SIP Trunk Side Interoperability Test Plans ( ) for detailed test cases. Feature Feature Description Issues Basic Call Automatic Call Distribution NuPoint Voic Packetization Personal Ring Groups Mobile Extension Teleworker Video Fax Making and receiving a call through the SKYPE SIP trunk, call holding, transferring, conferencing, busy calls, long calls durations, variable codec. Making calls to an ACD environment with RAD treatments, Interflow and Overflow call scenarios and DTMF detection. Terminating calls to a NuPoint voic boxes and DTMF detection. Forcing the 3300 MCD to stream RTP packets through its E2T card at different intervals, from 10ms to 80ms Receiving calls through SKYPE SIP trunk to a personal ring group. Also moving calls to/from the prime member and group members. Receiving a call through the SIP trunk to Mobile extensions and TUI interface. Also moving calls to/from Desktop and Twinned devices. Making and receiving a call through SKYPE SIP trunk to and from Teleworker extensions. Making and receiving a call through SKYPE SIP trunk with video capable devices. No video calls supported. T.38 and G711Fax Calls - No issues found - Issues found, cannot recommend to use - Issues found 2

7 Device Limitations and Known Issues This is a list of problems or not supported features when the SKYPE SIP trunk is connected to the Mitel Feature Private Extension Packetization rate RFC 2833 Long Ring Provisional Responses Video Fax Problem Description Outgoing call from Private Extension on the MCD will fail as a result of a 403 Forbidden message returned from SKYPE. Recommendation: Consult with SKYPE should support be required for this functionality. SKYPE does not support Packetization rate >= 60 ms. The MCD default packetization rate is 20ms and functions without issue and will not affect calls from the 3300 or any Mitel sets. This is simply another characteristic of this interop should any other SIP device require packetization rate greater than 50ms. Recommendation: Consult with SKYPE should support be required for this functionality. Sometimes outgoing call to PSTN network via SKYPE Service Provider lost RFC 2833 telephone event. Recommendation: Consult with SKYPE support regarding this functionality. SKYPE calls cannot ring for longer that 45 seconds, Calls will terminate after 45 seconds of ringing. Recommendation Consult with SKYPE should change of this functionality be required. SKYPE does not support Provision Responses methods. This will not affect general call performance but rather it is simply a characteristic of this interop. Recommendation: Consult with SKYPE should support be required for this functionality. Video calls is unavailable over the SKYPE SIP trunks. Recommendation: Consult with SKYPE should support be required for this functionality. SKYPE does not support T.38 faxing at this time, and Outgoing fax using G711 also fails. ( no fax transmission at all) Recommendation: Consult with SKYPE support regarding this functionality.

8 Network Topology This diagram shows how the testing network is configured for reference. Figure 1 Network Topology 4

9 Configuration Notes This section is a description of how the SIP Interop was configured. These notes should give a guideline how a device can be configured in a customer environment and how the SKYPE 3300 programming was configured in our test environment. Disclaimer: Although Mitel has attempted to setup the interop testing facility as closely as possible to a customer premise environment, implementation setup could be different onsite. YOU MUST EXERCISE YOUR OWN DUE DILIGENCE IN REVIEWING, planning, implementing, and testing a customer configuration MCD Configuration Notes The following steps show how to program a 3300 MCD to interconnect with the SKYPE. Network Requirements There must be adequate bandwidth to support the voice over IP. As a guide, the Ethernet bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice session (assumes 20ms packetization). As an example, for 20 simultaneous SIP sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and 0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any special engineering. Please refer to the 3300 Engineering guidelines for further information. For high quality voice, the network connectivity must support a voice-quality grade of service (packet loss <1%, jitter < 30ms, one-way delay < 80ms). Assumptions for the 3300 MCD Programming The SIP signaling connection uses UDP on Port 5060.

10 Licensing and Option Selection SIP Licensing Ensure that the 3300 MCD is equipped with enough SIP trunking licenses for the connection to the SKYPE. This can be verified within the License and Option Selection form. Enter the total number of licenses in the SIP Trunk Licences field. This is the maximum number of SIP trunk sessions that can be configured in the 3300 to be used with all service providers, applications and SIP trunking devices. Figure 2 License and Option Selection form 6

11 Class of Service Assignment The Class of Service Options Assignment form is used to create or edit a Class of Service and specify its options. Classes of Service, identified by Class of Service numbers, are referenced in the Trunk Service Assignment form for SIP trunks. Many different options may be required for your site deployment, but ensure that Public Network Access via DPNSS Class of Service Option is configured for all devices that make outgoing calls through the SIP trunks in the Busy Override Security set to Yes Campon Tone Security/FAX Machine set to Yes Public Network Access via DPNSS set to Yes

12 Figure 3 Class of Service form 8

13 Network Element Assignment Create a network element for a SIP Peer (SKYPE) as shown in Figure 4. Set the transport to UDP and port to Figure 4 Network Element Assignment

14 Network Element Assignment (Proxy) In addition, depending in your configuration, a Proxy may need to be configured to route SIP data to the service provider. If you have a Proxy server installed in your network, the 3300 MCD will require knowledge of this by programming the Proxy as a network element then referencing this proxy in the SIP Peer Profile assignment (later in this document). Figure 5 Network Element Assignment (Proxy) 10

15 Trunk Service Assignment This is configured in the Trunk Service Assignment form. The Trunk Service Assignment is defined for Trunk Service Number (41), which will be used to direct incoming calls to an answer point in the Program the Non-dial In or Dial In Trunks (DID) according to the site requirements and what type of service was ordered from your service provider. The figure below shows configuration for incoming DID calls. The 3300 will absorb the first 10 digits of the DID number from the SKYPE SIP Trunk leaving 4 digits for the 3300 to translate and ring the remaining 4 digit extension. For example, the SKYPE SIP Trunk delivers number to the The 3300 will absorb the first 10 digits ( ) leaving the 3300 to ring extension Extension 1219 must be programmed as a valid dialable number in the As an alternative way, you can create a System Speed Call number to associate the extension 1219 with the real telephone extension on 3300 ICP. Please refer to the 3300 System Administration documentation for further programming information. Figure 6 Trunk Service Assignment

16 SIP Peer Profile The recommended connectivity via SIP Trunking does not require additional physical interfaces. IP/Ethernet connectivity is part of the base 3300 MCD Platform. The SIP Peer Profile should be configured with the following options: Network Element: The selected SIP Peer Profile needs to be associated with previously created SKYPE Network Element. Registration User Name: Address Type: Enter the Use IP Address or FQDN in SIP messages. Maximum Simultaneous Calls: This entry should be configured to maximum number of SIP trunks provided by SKYPE. Outbound Proxy Server: Select the Network Element previously configured for the Outbound Proxy Server. SMDR Tag: If Call Detail Records are required for SIP Trunking, the SMDR Tag should be configured (by default there is no SMDR and this field is left blank). Trunk Service Assignment: Enter the trunk service assignment previously configured 41 in this configuration. Calling Line ID Options: The Default CPN (Calling Party Number) is applied to all outgoing calls unless there is a match in the "Outgoing DID Ranges" of the SIP Peer Profile. SKYPE could provide this number. Otherwise, use the one of DID numbers assigned on the trunk by the provider. Default CPN should be known to the provider s SIP switch otherwise it refuses to process any calls. Authentication Options: Since this SIP trunk is authenticated by IP address, leave this field blank. SDP Options: Currently (Mar 2010) video streams are not supported over SKYPE SIP trunk. Set option Allow Peer to Use Multiple Active M-Lines to No. Signaling and Header Manipulation Options: Set value for Session Timer to 0 because SKYPE does not support session timer. 12

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18 Figure 7 SIP Peer Profile Assignment 14

19 Digit Modification Number Ensure that Digit Modification for outgoing calls to SKYPE SIP Trunk absorbs or inject additional digits according to your dialling plan. In this test environment, we will be absorbing 2 digits (in this case you will need to dial 93 to access SKYPE SIP trunk; thus, digits 93 will be absorbed). Figure 8 Digit Modification Assignment

20 Route Assignment Create a route for SIP Trunks connecting a trunk to SKYPE SIP Trunk. In this test environment, the SIP trunk is assigned to Route Number 41. Choose SIP Trunk as a routing medium and choose the SIP Peer Profile and Digit Modification entry created earlier. Figure 9 SIP Trunk Route Assignment 16

21 ARS Digits Dialed Assignment ARS initiates the routing of trunk calls when certain digits are dialed from an extension. In this test environment, when a user dials 93, the call will be routed to SKYPE SIP Trunk (i.e. Route 41). Figure 11 ARS Digit Dialed Assignment

22 T.38 Fax Configuration Intra-zone 1 is used for fax over SKYPE SIP trunk because SKYPE does not support T38 at this time. Later if SKYPE can support T38, we can use the inter-zone 2 FAX profile. This form allows you to define the settings for FAX communication over the IP network. You can modify the default settings for the: Inter-zone FAX profile: defines the FAX settings between different zones in the network. There is only one Inter-zone FAX profile; it applies to all inter-zone FAX communication. It defaults to V.29, 7200bps. It defines the settings for FAX Relay (T.38) FAX communication. Intra-zone FAX profile: defines the FAX settings within each zone in the network. o o Profile 1 defines the settings for G.711 pass through communication. Profile 2 to 64 define the settings for FAX Relay (T.38) FAX communication. o All zones default to G.711 pass through communication (Profile 1). Figure 12 Fax Configuration 18

23 Zone Assignment By default, all zones are set to Intra-zone FAX Profile 1. Based on your network diagram, assign the Intra-zone FAX Profiles to the Zone IDs of the zones. If audio compression is required within the same zone, set Intra-Zone Compression to Yes. Figure 103 Zone Assignment The assignment of the pre-configured zone to the SIP Peer profile can be done in Network Element Assignment form.

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