Release Notes. Version 6.0
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1 Release Notes Version 6.0 Document #: LTRT February 2010
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3 SIP Release Notes Contents Table of Contents 1 What's New in Release Supported Hardware Platforms New Models and Hardware Configurations Introduced in this Release Existing Hardware Platforms Hardware Platforms No Longer Supported SIP New Features B2BUA Transcoding New Feature SIP Network Definitions Signaling Routing Domain (SRD) Entities SIP Interfaces Media Realm SIP Dialog Initiation Process Determining Source and Destination URL Source IP Group Classification IP-to-IP Routing IP-to-IP Inbound and Outbound Manipulation Media Handling Media Anchoring without Transcoding (Transparent) Media Anchoring with Transcoding Transcoding Modes Voice, RTP and RTCP New Features Security New Features PSTN New Features Web New Features SNMP New Features Miscellaneous New Features New Parameters SIP Parameters B2BUA Transcoding Parameters Voice, RTP and RTCP Parameters Security Parameters PSTN Parameters Existing ini File Parameters Now Configurable in the Web Modified Parameters SIP Parameters Voice, RTP and RTCP Parameters PSTN Parameters Obsolete Parameters Supported Features SIP Features Supported SIP Features Unsupported SIP Features SIP Compliance Tables SIP Functions SIP Methods SIP Headers SDP Headers SIP Responses Version February 2010
4 Mediant 2000 & Mediant PSTN to SIP Interworking Supported Interworking Features Interworking Features Not Supported DSP Firmware Templates Template Mix Feature Product Capability Comparison Known Constraints Voice, RTP and RTCP Constraints Infrastructure Constraints Networking Constraints Security Constraints High Availability Constraints PSTN Constraints Web Constraints SNMP Constraints CLI Constraints Resolved Constraints Voice, RTP and RTCP Resolved Constraints Web Resolved Constraints SNMP Resolved Constraints Earlier Releases SIP Release Notes 4 Document #: LTRT-69017
5 SIP Release Notes Contents List of Tables Table 1-1: New SIP Parameters for Release Table 1-2: New B2BUA Transcoding Parameters for Release Table 1-3: New Voice, RTP and RTCP Parameters for Release Table 1-4: New Security Parameters for Release Table 1-5: New PSTN Parameters for Release Table 1-6: ini File Parameters now Configurable in the Web Interface for Release Table 1-7: Modified SIP Parameters for Release Table 1-8: Modified Voice, RTP and RTCP Parameter for Release Table 1-9: Modified PSTN Parameter for Release Table 1-10: Obsolete Parameters Table 2-1: Supported SIP Functions Table 2-2: Supported SIP Methods Table 2-3: Supported SIP Headers Table 2-4: Supported SDP Headers Table 2-5: Supported 1xx SIP Responses Table 2-6: Supported 2xx SIP Responses Table 2-7: Supported 3xx SIP Responses Table 2-8: Supported 4xx SIP Responses Table 2-9: Supported 5xx SIP Responses Table 2-10: Supported 6xx SIP Responses Table 2-11: DSP Firmware Templates for Mediant Table 2-12: DSP Firmware Templates for Mediant Table 2-13: DSP Firmware Templates for Mediant E1 /21 T Table 2-14: Template Mix Feature Channel Capacity Table 2-15: Capability Comparison between Devices Version February 2010
6 Mediant 2000 & Mediant 3000 Reader s Notes SIP Release Notes 6 Document #: LTRT-69017
7 SIP Release Notes Notices Notice This document describes the release of the AudioCodes Mediant 2000 Voice-over-IP (VoIP) SIP media gateway (housing the TP-1610 blade) and Mediant 3000 VoIP SIP media gateway (housing the TP-6310 or TP-8410 TrunkPack (TP) SIP cpci blades). Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions. Updates to this document and other documents can be viewed by registered customers at Copyright 2010 AudioCodes Ltd. All rights reserved. This document is subject to change without notice. Date Published: February Trademarks AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect, VoIPerfectHD, What s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners. Product specifications are subject to change without notice. WEEE EU Directive Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed of with unsorted waste. Please contact your local recycling authority for disposal of this product. Customer Support Customer technical support and service are provided by AudioCodes Distributors, Partners, and Resellers from whom the product was purchased. For Customer support for products purchased directly from AudioCodes, contact support@audiocodes.com. Abbreviations and Terminology Each abbreviation, unless widely used, is spelled out in full when first used. Only industrystandard terms are used throughout this manual. Hexadecimal notation is indicated by 0x preceding the number. Version February 2010
8 Mediant 2000 & Mediant 3000 Related Documentation Document Name Product Reference Manual for SIP CPE Devices Mediant 2000 SIP Installation Manual Mediant 2000 SIP User's Manual Mediant 3000 SIP User's Manual Mediant 3000 & IPmedia 3000 SIP-MGCP-MEGACO Installation Manual CPE SIP Configuration Guide for IP Voice Mail Notes: Throughout this manual, unless otherwise specified, the term device refers to the Mediant 2000 and Mediant 3000 media gateways. SIP Release Notes 8 Document #: LTRT-69017
9 SIP Release Notes 1. What's New in Release What's New in Release 6.0 Note: This document uses a one-row table with check boxes convention to indicate the products for which each feature is applicable. Only the products with marked check boxes are applicable to the feature. For example, the table below indicates that the feature is applicable only to Mediant 3000 High Availability (HA) housing the TP Mediant 2000 Mediant 3000 Mediant Supported Hardware Platforms New Models and Hardware Configurations Introduced in this Release Not applicable Existing Hardware Platforms The following existing hardware platforms are supported in this release: Mediant 3000 system: 2U carrier-grade chassis, supporting multiple configuration options of VoP cpci blades: Mediant 3000 hosting a single TP-8410 blade, providing 16 E1/ 21 T1 PSTN interfaces. Mediant 3000 hosting a single TP-8410 blade, providing up to 63 E1/ 84 T1 PSTN interfaces. Mediant 3000 hosting two TP-8410 blades for 1+1 High Availability (HA), providing up to 16 E1/ 21 T1 PSTN interfaces. Mediant 3000 hosting two TP-8410 blades for 1+1 HA, providing up to 63 E1/ 84 T1 PSTN interfaces. Mediant 3000 hosting a single TP-6310 blade, providing SONET/SDH or T3 PSTN interfaces. Mediant 3000 hosting two TP-6310 blades for 1+1 HA, providing SONET/SDH or T3 PSTN interfaces. Mediant 3000 hosting a single TP-8410 blade providing 16 E1/ 21 T1 PSTN interfaces with an integrated CPU (Intel Pentium) blade (M3K-ICPU-1), hosting third-party applications (such as SS7 GWC). Mediant 3000 hosting a single TP-8410 blade providing up to 63 E1/ 84 T1 PSTN interfaces with an integrated CPU (Intel Pentium) blade (M3K-ICPU-1), hosting third-party applications (such as SS7 GWC). Mediant 2000 system: 1U chassis hosting a TP-1610 cpci blade with E1/T1 interfaces. Version February 2010
10 Mediant 2000 & Mediant Hardware Platforms No Longer Supported Not applicable. 1.2 SIP New Features The device supports the following new SIP features: 1. Sending SIP Response to UDP Port from where SIP Request Received regardless of "rport" Parameter: Mediant 2000 Mediant 3000 Mediant 3000 In previous releases, the device sent SIP responses to the UDP port defined in the SIP Via header. If the Via header contained the "rport" parameter, the device sent the response to the UDP port from where the SIP request was received. In this release, the device can be configured (using the new parameter, SIPForceRport) to send SIP responses to the UDP port from where the SIP request was received even if the "rport" parameter is not received in the Via header. Relevant parameter: SIPForceRport. 2. Maximum Proxy Sets Increased to 10: Mediant 2000 Mediant 3000 Mediant 3000 The device now allows you to configure up to 10 Proxy Sets (compared to only 6 in previous releases). These are configured in the 'Proxy Sets' table (ProxySet). Relevant Parameter: ProxySet. 3. Offered Coders Increased to 10: Mediant 2000 Mediant 3000 Mediant The device now supports the configuration of up to 10 coders (compared to only 5 in the previous release) for offering the remote end. This also applies to Coder Groups, where up to 10 coders can now be defined per Coder Group (compared to only 5 in the previous release). In addition, a new parameter, CodersGroup now replaces the CoderName parameter (from previous versions). This new parameter supports backward compatibility, allowing users from previous versions to seamlessly upgrade to Version 6.0 (the coders defined under the CoderName parameter are transferred to the CodersGroup parameter). Relevant parameter: CodersGroup. SIP Release Notes 10 Document #: LTRT-69017
11 SIP Release Notes 1. What's New in Release 6.0 Parameters Added to Tel Profile: Mediant 2000 Mediant 3000 Mediant 3000 The following parameters have now been added to the device's Tel Profile feature: 5. SwapTelToIpPhoneNumbers - representing the global parameter SwapTEl2IPCalled&CallingNumbers EnableAGC - representing the global parameter EnableAGC ECNlpMode - representing the global parameter ECNlpMode Relevant parameter: TelProfile. Trunk Groups Increased to 120: Mediant 2000 Mediant 3000 Mediant The device now supports the configuration of up to 120 Trunk Groups (compared to 24 for Mediant 2000, and 100 for Mediant 3000 in previous releases). The Trunk Group ID range has also been increased accordingly from 0-99 to This increase is reflected in the Trunk Group (TrunkGroup) and Trunk Group Settings (TrunkGroupSettings) tables. Relevant parameters: TrunkGroup; TrunkGroupSettings. Trunk and Channel Cyclic Ascending for Channel Select Mode: Mediant 2000 Mediant 3000 Mediant The device now supports a new method - Trunk and Channel Cyclic Ascending - for allocating incoming IP-to-Tel calls to a trunk s B-channels. This method combines two existing methods: Cyclic Ascending and Trunk Cyclic Ascending. This method selects the next physical trunk (pertaining to the Trunk Group) and then selects the B-channel of this trunk according to the cyclic ascending method (i.e., selects the channel after the last allocated channel). For example, assume that a Trunk Group includes two physical trunks, 0 and 1: a. For the first incoming call, the first channel of Trunk 0 is allocated. b. For the second incoming call, the first channel of Trunk 1 is allocated. c. For the third incoming call, the second channel of Trunk 0 is allocated. Relevant parameters: ChannelSelectMode; TrunkGroupSettings. Version February 2010
12 Mediant 2000 & Mediant 3000 Tel-to-IP Destination Number Manipulation Entries Increased to 120: Mediant 2000 Mediant 3000 Mediant The device now allows you to configure up to 120 Tel-to-IP destination number manipulation rules (compared to 100 in the previous release). These are configured in the 'Destination Phone Number Manipulation Table for Tel to IP Calls' table (NumberMapTel2IP). Relevant parameter: NumberMapTel2IP. IP-to-Tel Redirect Number Manipulation: Mediant 2000 Mediant 3000 Mediant The device now supports IP-to-Tel redirect number manipulation, configured using the new table, Redirect Number IP to Tel table. This feature allows you to manipulate the value of the received SIP Diversion, Resource-Priority, or History-Info header, which is then added to the Redirecting Number Information Element (IE) in the ISDN Setup message sent to the Tel side. Relevant parameter: RedirectNumberMapIp2Tel. Tel-to-IP Redirect Number Manipulation: Mediant 2000 Mediant 3000 Mediant The device now supports Tel-to-IP Redirect Number manipulation, configured using the new table, Redirect Number Tel to IP table. This feature allows you to manipulate the prefix of the redirect number received from the PSTN for the outgoing SIP Diversion, Resource-Priority, or History-Info header that is sent to IP. Relevant parameter: RedirectNumberMapTel2Ip. Routing IP-to-Tel Calls to Specific Trunk Groups According to CIC Parameter in Request URI: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports IP-to-Tel routing decisions based on the SIP carrier identification code ("cic") parameter. It uses the "cic" parameter in the incoming SIP INVITE message to route the call to a specific Trunk Group. The SIP carrier identification code (CIC) enables the transmission of the CIC parameter from the SIP network to the ISDN (CIC=+<country code of carrier><cic of preferred interlata carrier of caller>). The CIC parameter is a three- or four- digit code used in routing tables to identify the network that serves the remote user when a call is routed over many different networks. The CIC parameter is carried in SIP INVITE and maps to the ISDN Transit Network Selection Information Element (TNS IE) in the outgoing ISDN Setup message (if the EnableCIC parameter is set to 1). The TNS IE identifies the requested transportation networks and allows different providers equal access support, based on customer choice. SIP Release Notes 12 Document #: LTRT-69017
13 SIP Release Notes 1. What's New in Release 6.0 For supporting this new feature, this release introduces the new parameter, AddCicAsPrefix. When this parameter is enabled, the device adds the "cic" prefix to the destination number (for IP-to-Tel calls). For example: INVITE SIP/ After number manipulation performed by the device, the destination number results in "cic ". Note: After the cic prefix is added, the IP-to-Trunk Group Routing table can be used to route this call to a specific Trunk Group. The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to the ISDN. Relevant parameter: AddCicAsPrefix. SIP "dtg" Parameter for Routing IP-to-Tel Calls to Trunk Groups: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports the "dtg" parameter for defining the Trunk Group for routing IP-to-Tel calls. This parameter can be used instead of the "tgrp/trunk-context" parameters. The "dtg" parameter appears in the INVITE, for example: INVITE sip:123456@ ;dtg=56;user=phone SIP/ The "dtg" parameter also appears in the SIP To header. This feature is enabled by the new parameter, UseBroadsoftDTG (set to 1). If the Trunk Group is not found based on the "dtg" parameter, the IP to Trunk Group Routing table is used instead for routing the call to the appropriate Trunk Group. Relevant parameter: UseBroadsoftDTG. IP-to-Tel Routing Precedence using "tgrp"/"dtg" Parameters or IP to Trunk Group Table: Mediant 2000 Mediant 3000 Mediant 3000 In previous releases, IP-to-Tel routing was determined by the IP to Trunk Group Routing table (PSTNPrefix ini file parameter), and only if a matching rule was not found in this table did the device use the "tgrp"/"dtg" parameters for routing the call. However, in this release, you can change this priority so that the device first places precedence on the tgrp/dtg parameters for IP-to-Tel routing. If the received INVITE request URI does not contain the tgrp/dtg parameters, or if the Trunk Group number is not defined, then the IP to Trunk Group Routing table is used for routing the call. The IP-to-Tel Routing Precedence feature is enabled using a new parameter, TGRProutingPrecedence. If set to 1, the device performs routing according to the tgrp/dtg parameters. If set to 0 (default), the behavior is the same as in previous releases (first locates a match in the routing table and only if not found, attempts to route the call according to the tgrp parameter). Version February 2010
14 Mediant 2000 & Mediant 3000 Below is an example of an INVITE request URI with the tgrp parameter, indicating that the IP call should be routed to Trunk Group 7: INVITE SIP/ Note that the UseSIPTgrp parameter must be set to 2 for enabling routing based on the SIP tgrp parameter. Relevant parameters: UseSIPTgrp; TGRProutingPrecedence. IP-to-Tel Call Forwarding to IP Destination upon Busy Trunk Group: Mediant 2000 Mediant 3000 Mediant The device now supports the forwarding of IP-to-Tel calls to a different IP destination, using SIP 3xx response if a Trunk Group has no free channels (i.e., busy Trunk Group). This feature is configured using the new table, Forward On Busy Trunk Destination, which defines an alternative IP destination (IP address, port and transport type) per Trunk Group. The device forwards calls using this new table only if no alternative IP-to-Tel routing has been configured or alternative routing fails, and one of the following reasons (included in the SIP Diversion header of 3xx messages) exists: out-of-service - all trunks are unavailable/disconnected "unavailable" - all trunks are busy or unavailable Relevant parameter: ForwardOnBusyTrunkDest. Selecting SIP Header for IP-to-Tel Destination Number: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports selecting the SIP header for obtaining the called (destination) number (for IP-to-Tel calls). The device can be configured, using the new parameter SelectSourceHeaderForCalledNumber (replacing now obsolete IsUseToHeaderAsCalledNumber parameter), to use one of the following headers for obtaining the destination number: Request-URI (default) To P-Called-Party-ID Relevant parameters: SelectSourceHeaderForCalledNumber. SIP Release Notes 14 Document #: LTRT-69017
15 SIP Release Notes 1. What's New in Release LDAP Support for Microsoft Active Directory Queries: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports Lightweight Directory Access Protocol (LDAP), allowing the device to make call routing decisions based on information stored on a third-party LDAP server (or Microsoft s Active Directory-based enterprise directory server). Therefore, this feature enables the usage of one common, popular database to manage and maintain information regarding user s availability, presence, and location. The LDAP feature can be configured using the ini file, Web interface, SNMP, and CLI (for debugging only). The device supports 20 for Mediant 2000 and 80 for Mediant 3000 LDAP search requests. Connection: The device connects and binds to the remote LDAP server either during the service s initialization (at device start-up) or whenever the LDAP server's IP address and port is changed. Service makes 10 attempts to connect and bind to the remote LDAP server with a timeout of 20 seconds between attempts. If connection fails, the service remains in disconnected state until either the LDAP server's IP address or port is changed. If connection to the LDAP server later fails, the service attempts to reconnect, as described previously. The SNMP alarm acldaplostconnection is sent when connection is broken. Upon successful reconnection, the alarm is cleared. Binding to the LDAP server can be anonymous or not. For anonymous binding, the LDAPBindDN and LDAPPassword parameters must not be defined or set to an empty string. The address of the LDAP server can be a DNS name (using the LDAPServerName parameter) or an IP address (using the LDAPServerIP parameter). Search: To run a search using the LDAP service, the path to the directory s subtree where the search is to be performed must be defined (using the LDAPSearchDN parameter). In addition, the search key (known as filter in LDAP references), which defines the exact DN to be found and one or more attributes whose values should be returned, must be defined. If connection to the LDAP server is disrupted during the search, all search requests are dropped and an alarm indicating a failed status is sent to client applications. Version February 2010
16 Mediant 2000 & Mediant CLI: The LDAP CLI is located in the directory IPNetworking\OpenLdap. The following commands can be used: LdapSTatus - displays connection status LdapSearch - searches an LDAP server LDapOpen - opens connection to the LDAP server using parameters provided in configuration file LDapSetDebugmode - sets the LdapDebugLevelMode parameter LDapGetDebugmode gets the LdapDebugLevelMode parameter value Relevant parameters: LDAPServiceEnable; LDAPServerIP; LDAPServerDomainName; LDAPServerPort; LDAPPassword; LDAPBindDN; LDAPSearchDN; LDAPDebugMode; LDAPServerMaxRespondTime. Active Directory-Based Tel-to-IP Call Routing for Microsoft OCS 2007 Environment: Mediant 2000 Mediant 3000 Mediant 3000 Typically, enterprises wishing to deploy Microsoft s Office Communication Server 2007 (OCS 2007) are faced with a complex, call routing dial plan when migrating users from their existing PBX/IP-PBX to the OCS 2007 platform. As more and more end-users migrate to the new voice system, dialing plan management and PBX link capacity can be adversely impacted. Moreover, it s easy to perceive that even a temporary failure (or disconnection) of Microsoft s Office Communications Server 2007 Mediation Server (Mediation Server) results in no incoming voice calls from the PBX/IP-PBX/PSTN and therefore, it will be impossible to reach the user on the user s Microsoft Office Communicator (OC) client. This new feature enables the device to make Tel-to-IP call routing decisions based on information stored on Microsoft s Active Directory-based (AD) enterprise directory server. Therefore, this implements one common, central database to manage and maintain information regarding user s availability, presence, and location. Based on queries sent to the AD, this feature allows you to route incoming Tel calls to one of the following IP domains: PBX/IP-PBX (for users yet to migrate to the OCS 2007 platform) OCS clients (clients connected to the OCS 2007 platform) Mobile The device queries the AD using the destination number. The device's AD queries returns up to three user phone number IP destinations, each pertaining to one of the IP domains listed above. The device routes the call according to the following priority: OCS SIP address: Call is routed to Mediation Server (which then routes the call to the OCS client) Mobile number: If the Mediation Server or OCS client is unavailable (e.g., SIP response 404 "Not Found" upon INVITE sent to OCS client), the device routes the call to the user's mobile number (if exists in the AD). PBX/IP-PBX number: If no OCS client exits in the AD, then the device routes the call to the PBX/IP-PBX (if this fails, the call is routed to the mobile number, if exists). SIP Release Notes 16 Document #: LTRT-69017
17 SIP Release Notes 1. What's New in Release 6.0 For enterprises implementing a PBX/IP-PBX system but yet to migrate to the OCS 2007 platform, if the PBX/IP-PBX system is unavailable, the device queries the AD for the users mobile phone number and then routes the call through the PSTN to the mobile destination. This feature is configured in the Outbound IP Routing table, where the "LDAP" keywords are entered for the destination phone prefixes. For each IP domain (listed above), the destination numbers will be prefixed (case-sensitive) as follows: OCS client number: "OCS:" PBX number: "PBX:" Mobile number: "MOBILE:" LDAP failure: "LDAP_ERR:" Note that these prefixes are only involved in the routing and manipulation stages; they are not used as the final destination number. In addition, once you have configured the LDAP parameters (see previous feature), you need to enter the "LDAP" value for the destination IP address of the LDAP server in the Outbound IP Routing table. For enabling alternative routing, you need to enable the alternative routing mechanism and configure corresponding SIP reasons for alternative routing. For this feature, alternative routing always starts again from the top of the table (first routing rule entry) and not from the next row. This feature introduces three new parameters to configure the attribute names in the Active Directory used in the LDAP query: AD attribute for Mediation Server: MSLDAPOCSNumAttributeName (the default is "msrtcsipprimaryuseraddress") AD attribute for PBX/IP-PBX: MSLDAPPBXNumAttributeName (the default is "telephonenumber") AD attribute for mobile number: MSLDAPMobileNumAttributeName (the default is "mobile") Below is an example of configuring Active Directory-based routing rules in the Outbound IP Routing Table: First rule: sends call to IP-PBX ( ) if AD query replies with prefix "PBX:" Second rule: sends call to OCS client (i.e., Mediation Server at ) if AD query replies with prefix "OCS:" Third rule: sends call to users mobile phone number (to PSTN through the device's IP address, ) if AD query replies with prefix "MOBILE:" Version February 2010
18 Mediant 2000 & Mediant Fourth rule: sends call to IP address of device (e.g., ) if no response from LDAP server Fifth rule: sends query of received Tel destination number to LDAP server and then routes the call according to query reply and routing rules at top of table Sixth rule: if LDAP functionality is not enabled, routes calls to IP address Therefore, once the device receives the incoming Tel call, the first rule that it uses is the fifth rule, which queries the AD server. When the AD replies, the device searches the table, from the first rule down for the matching destination phone prefix (i.e., "PBX:", "OCS:", "MOBILE:", and "LDAP_ERR:"), and then sends the call to the appropriate destination. Relevant parameters: MSLDAPOCSNumAttributeName; MSLDAPPBXNumAttributeName; MSLDAPMobileNumAttributeName. MLPP Preemption Events in SIP Reason Header - "preemption; cause=4" Instead of "preemption; cause=3": Mediant 2000 Mediant 3000 Mediant Preemption ensures that existing calls of lower priority are terminated in order to accept higher priority calls. The SIP Reason header indicates the reason a preemption event occurred and the type of preemption event. Non-IP Preemption indicates that the session preemption has occurred in a non-ip portion of the infrastructure, and this is the Reason cause code given by the SIP device. In the previous release, the device sent a SIP BYE or CANCEL request, or 480, 486, 488 responses (as appropriate) with a Reason header whose reason-params contained the value preemption; cause=3; text=" Generic Preemption. However, in this release, according to Unified Capabilities Requirements 2008 (UCR 2008), the Reason header instead includes Reason: preemption ;cause=4 ;text= Non-IP Preemption, in the following scenarios: Whenever the device performs a network preemption of a busy call (when a high priority call is received), the device sends a SIP BYE or CANCEL request with a Reason header whose Reason-params has the value preemption ;cause=4 ;text= Non-IP Preemption, instead of cause=3. Whenever the device performs a preemption of a B-channel for a Tel-to-IP outbound call request from the softswitch for which it has not received an answer response (e.g., Connect), then the following sequence of events occurs: a. The device sends a Q.931 DISCONNECT over the ISDN MLPP PRI to the partner switch to preempt the remote end instrument. b. The device sends a 488 (Not Acceptable Here) response that includes a Reason header whose Reason-params contains the value preemption; cause=4; text= Non-IP Preemption (instead of cause=3). SIP Release Notes 18 Document #: LTRT-69017
19 SIP Release Notes 1. What's New in Release 6.0 SIP Reason Header (Reason: preemption; cause=5; text= Network Preemption ) for MLPP Preemption Network Events (RFC 4411): Mediant 2000 Mediant 3000 Mediant 3000 The device now supports the SIP Reason header (as defined by RFC 4411) to support the signaling of preemption-related events in the network. A new preemption cause class is defined along with five cause values for the new cause class. Reason: preemption ;cause=1 ;text= UA Preemption Reason: preemption ;cause=2 ;text= Reserved Resources Preempted Reason: preemption ;cause=3 ;text= Generic Preemption Reason: preemption ;cause=4 ;text= Non-IP Preemption Reason: preemption; cause=5; text= Network Preemption Within the DSN network, the following SIP request messages and response codes for specific call scenarios have been identified for signaling this new class of preemption causes: SIP:BYE - If an active call is being preempted by another call CANCEL - If an outgoing call is being preempted by another call (Temporarily Unavailable), 486 (User Busy), 488 Not Acceptable Here) - Due to incoming calls being preempted by another call The device receives SIP requests with preemption reason cause=5 in the following cases: 2. Whenever the softswitch performs a network preemption of an active call, the following sequence of events occurs: a. The softswitch sends the device a SIP BYE request with a Reason header whose Reason-params contain the value preemption; cause=5; text= Network Preemption. b. The device initiates the release procedures for the B-channel associated with the call request and maps the preemption cause to PRI Cause = #8 Preemption. This value indicates that the call is being preempted. For PRI, it also indicates that the B-channel is not reserved for reuse. c. The device sends a SIP 200 OK in response to the received BYE, before the SIP end instrument can proceed with the higher precedence call. Whenever the softswitch performs a network preemption of an outbound call request for the device that has not received a SIP 2xx response, the following sequence of events occur: a. The softswitch sends the device a SIP 488 (Not Acceptable Here) response code with a Reason header whose Reason-params contain the value preemption;cause=5; text= Network Preemption. The device initiates the release procedures for the B-channel associated with the call request and maps the preemption cause to PRI Cause = #8 Preemption. b. The device deactivates any user signaling (e.g., ringback tone), and when the call is terminated, it sends a SIP ACK message to the softswitch. Version February 2010
20 Mediant 2000 & Mediant Same SIP Resource-Priority Header in Subsequent Re-INVITEs for MLPP: Mediant 2000 Mediant 3000 Mediant 3000 The device now sends a 403 Forbidden response upon receipt of a Re-INVITE whose Resource-Priority header is different to that received in the initial INVITE. This is for MLPP calls in the IP-to-Tel direction. 20. Invalid SIP Resource-Priority Header for MLPP: Mediant 2000 Mediant 3000 Mediant 3000 When the device receives an unknown (invalid) Network-Domain in the SIP Resource- Priority header, the Network-Domain is changed to "dsn" and the r-priority is set to 0. The precedence-domain remains as the original. In other words, the device sends the following namespace: dsn-<original precedence-domain> Warning Header Included in 488 Response for Rejected MLPP Calls: Mediant 2000 Mediant 3000 Mediant The device now rejects an MLPP call if a B-channel is unavailable to preempt, by sending a 488 (Not Acceptable Here) response code that includes a Warning header field with warning code 370 (Insufficient Bandwidth) to the softswitch serving the SIP entity. This warning code is added to the Warning header (and the MLPP call is subsequently rejected) in the following cases: If there are no preemptable resources to accommodate the new outbound precedence call request. Whenever the device performs a preemption of a B-channel for an outbound call (initiated from IP side) request for which it has not received a response (e.g. Connect) from the Tel side, then the following sequence occurs: a. The device sends an Q.931 DISCONNECT over the ISDN. b. The device sends to the IP side a 488 (Not Acceptable Here) response code that includes the Warning header field with the warning code. After sending the Q.931 SETUP related to the preempting call, the switch can reject the call by sending in a clearing message a cause of Precedence Call Blocked (Cause Value #46). Upon receipt of this cause, the device sends a 488 (Not Acceptable Here) response code that includes a Warning header field with warning code 370 (Insufficient Bandwidth) in order to perform precedence failure intercept announcement treatment. Note: When preempting a channel on ALERT state, the device rejects the MLPP call by sending a 480 response (not 488 and no Warning header is sent). SIP Release Notes 20 Document #: LTRT-69017
21 SIP Release Notes 1. What's New in Release 6.0 Configurable MLPP Network Identifier: Mediant 2000 Mediant 3000 Mediant 3000 The device now allows you to configure the MLPP network identifier (for IP-to-ISDN calls), according to the UCR 2008 and ITU Q.955 specifications. This specifies that octets 5 and 6 of the MLPP ISDN PRI Precedence Level Information Element should represent the network identifier (international prefix). In previous releases, this was hard-coded to USA Telephony Country Code (TCC) 1. This network identifier is included in the Facility IE of the ISDN Setup message. The network identifier can be defined using the new parameter MLPPNetworkIdentifier. For example: MLPPNetworkIdentifier set to default (i.e., USA, 1): PlaceCall- MLPPNetworkID:0100 MlppServiceDomain:123abc, MlppPrecLevel:5 Fac(1c): 91 a d 0a a a bc MLPPNetworkIdentifier set to 972: PlaceCall- MLPPNetworkID:7209 MlppServiceDomain:123abc, MlppPrecLevel:5 Fac(1c): 91 a d 0a a a bc MLPPNetworkIdentifier set to 490: PlaceCall- MLPPNetworkID:9004 MlppServiceDomain:123abc, MlppPrecLevel:5 Fac(1c): 91 a a d 0a a a bc Relevant parameter: MLPPNetworkIdentifier. 23. DSCP Based on Resource-Priority Header upon Receipt of SIP UPDATE: Mediant 2000 Mediant 3000 Mediant 3000 Upon receipt of the SIP UPDATE (with or without SDP), the device populates the Differentiated Service Code Point (DSCP) markings in the session media stream packets, based on the received precedence level from the SIP Resource-Priority header. 24. Version February 2010
22 Mediant 2000 & Mediant 3000 Enhanced Call Forking Support: Mediant 2000 Mediant 3000 Mediant 3000 The device now allows the configuration of a timeout (in seconds) that is started once the first SIP 2xx response has been received for a User Agent when a proxy server performs call forking (proxy server forwards the INVITE to multiple SIP User Agents). The device sends a SIP ACK and BYE in response to any additional SIP 2xx received from the proxy within this timeout. Once this timeout elapses, the device ignores subsequent SIP 2xx responses. In addition, the number of supported forking calls per channel has been increased from 4 to 20. In other words, the device can now receive up to 20 forking responses from a single INVITE message. Relevant parameter: ForkingTimeOut. 25. Fake Retry-After Header: Mediant 2000 Mediant 3000 Mediant This feature enables the device to operate with proxy servers that do not include the Retry-After SIP header in SIP 503 (Service Unavailable) responses to indicate an unavailable service. The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting SIP client. The value of this field can either be an HTTP-date or an integer number of seconds (in decimal) after the time of the response. The device maintains a list of available proxies, by using the Keep-Alive (KA) mechanism. The device checks the availability of proxies by sending SIP OPTIONS every KA timeout to all potential proxies. However, some third-party media servers reply to SIP OPTIONS even if they are unavailable. In such cases, the third-party server rejects the SIP INVITE, by sending a 503 (Service Unavailable) response. As a result, the device performs a failover and must periodically retry the availability of the server, by sending new calls to it to detect the possibility that the anomaly condition has been cleared. In previous releases, upon receipt of a SIP 503 response, the device discarded the call and the proxy remained in the live proxy list, since it responded to the device's SIP OPTIONS. Unless the 503 response included a Retry-After response-header, the device did not send new call to the proxy for a period specified in the header. Therefore, for third-party media servers that do not support the Retry-After responseheader, this release introduces a new parameter, FakeRetryAfter to resolve this issue. If this parameter is set to a positive value (in seconds), when the device receives a 503 response without a Retry-After response-header, it behaves as if the 503 response included a Retry-After response-header with the period specified by this parameter. If this parameter is set to zero, this "Fake Retry-After" feature is disabled. Relevant parameter: FakeRetryAfter. SIP Release Notes 22 Document #: LTRT-69017
23 SIP Release Notes 1. What's New in Release 6.0 Increased Number of SIP URIs in Received 302 Contact Header: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports the receipt of up to eight SIP Uniform Resource Identifiers (URIs) in the received 302 Contact header. This feature allows the device to handle the received redirection (302) response messages from the proxy with one or more contacts in one or more Contact headers (for example, Contact: The device uses the URIs in the Contact header, which could be one per Contact header or one Contact header could have multiple URIs, to formulate one or more new outbound call requests. 27. Early SIP 183 Response upon Receipt of INVITE: Mediant 2000 Mediant 3000 Mediant The device now supports sending a SIP 183 Session Progress response with SDP immediately upon receipt of an INVITE message. However, the device sends the RTP packets only after it receives an ISDN Progress, Alerting with Progress indicator, or Connect message from the PSTN. This feature is enabled by the new parameter, EnableEarly183 (when set to 1), and also by the existing parameter, EnableEarlyMedia (when set to 1). For example, if the device receives an ISDN Progress message, it starts sending RTP packets according to the initial negotiation without having to send the 183 response again. Therefore, this feature reduces clipping of early media. Relevant parameters: EnableEarly183; EnableEarlyMedia. Mapping Additional SIT Tones to Q.850 Causes: Mediant 2000 Mediant 3000 Mediant 3000 Until now, the device was capable of detecting and reporting the following Special Information Tones (SIT) types from the PSTN: SIT-NC (No Circuit found) SIT-IC (Operator Intercept) SIT-VC (Vacant Circuit - non-registered number) SIT-RO (Reorder - System Busy) These four SIT tones were mapped to Q.850 cause, using the SITQ850Cause parameter (set to 34, by default). In this release, the device now also supports the detection of an additional three SIT tones (which are detected as one of the above SIT tones): The NC* SIT tone - detected as NC The RO* SIT tone - detected as RO The IO* SIT tone - detected as VC Version February 2010
24 Mediant 2000 & Mediant 3000 The device can now map each of these SIT tones to a Q.850 cause and then map them to SIP 5xx/4xx responses, using the parameters SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC, and SITQ850CauseForRO. Note that if these parameters are not used (default), the SIT specific tone is mapped according to the configuration of the SITQ850Cause parameter. The SIT tones and their frequency durations reported by the device are shown in the table below: Special Information Tones (SITs) Name Description First Tone Frequency Duration Second Tone Frequency Duration Third Tone Frequency Duration (Hz) (ms) (Hz) (ms) (Hz) (ms) NC No circuit found IC Operator intercept VC Vacant circuit (non registered number) RO Reorder (system busy) NC* RO* IO* Relevant parameters: SITQ850CauseForNC; SITQ850CauseForIC; SITQ850CauseForVC; SITQ850CauseForRO; SITQ850Cause. 29. IP-to-IP RTP Tunneling: Mediant 2000 Mediant 3000 Mediant 3000 The device now supports IP-to-IP RTP tunneling. This feature is relevant only when transcoding is not required in the IP-to-IP path, and instead, a UDP-to-UDP path is used to forward RTP packets internally. In this mode, RTP events (i.e., Broken Connection, First Incoming Packet, and RTCP counters) are not generated. 30. Transcoding using Third-Party Call Control (RFC 4117): Mediant 2000 Mediant 3000 Mediant 3000 The device now supports RFC Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc) - providing transcoding services (i.e., acting as a transcoding server). This is implemented in scenarios where two SIP User Agents (UA) wish to establish a session, but do not have a common coder or media type. When such incompatibilities exist, the UAs need to invoke transcoding services to successfully establish the session. (Note that transcoding can also be performed using NetAnn according to RFC 4240.) To enable the RFC 4117 feature, this releases introduces a new parameter, EnableRFC4117Transcoding, which must be set to 1 (after which a device reset is required). SIP Release Notes 24 Document #: LTRT-69017
25 SIP Release Notes 1. What's New in Release 6.0 The 3pcc call flow is as follows: The device receives from one of the UAs, a single INVITE with an SDP containing two media lines. Each media represents the capabilities of each of the two UAs. The device needs to find a match for both of the media, and opens two channels with two different media ports to the different UAs. The device performs transcoding between the two voice calls. In the example below, the Application Server sends a special INVITE that consists of two media (m=) lines to perform transcoding between G.711 and G.729. m=audio RTP/AVP 0 c=in IP4 A.example.com m=audio RTP/AVP 18 c=in IP4 B.example.com Relevant parameter: EnableRFC4117Transcoding. 31. Forced Expiration (SIP Unregistration) using Contact Header Value "*": Mediant 2000 Mediant 3000 Mediant 3000 The device now supports the removal of SIP UA registration bindings in a Registrar, according to RFC Registrations are soft state and expire unless refreshed, but can also be explicitly removed. A client can attempt to influence the expiration interval selected by the registrar. A UA requests the immediate removal of a binding by specifying an expiration interval of "0" for that contact address in a REGISTER request. UAs should support this mechanism so that bindings can be removed before their expiration interval has passed. The REGISTER-specific Contact header field value of "*" applies to all registrations, but it can only be used if the Expires header field is present with a value of "0". Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record (AOR) without knowing their precise values. This feature is supported by the introduction of the new parameter, UnRegistrationMode. Relevant parameter: UnregistrationMode. 32. Hiding SIP Passwords: Mediant 2000 Mediant 3000 Mediant 3000 The device now hides configured SIP passwords. Passwords are configured in the 'Proxy & Registration' and 'Account Table' pages. Once you configure a password in the Web interface (and the Submit button is clicked), the Web GUI displays the entered password as an asterisk (*). If you load an ini file that includes a configured password, the Web GUI also displays it as an asterisk. When you save an ini file to a PC, the global parameter Password and its value are not displayed in the file. If a password is defined in a table ('Account Table'), the saved ini file displays the password value as an asterisk. Note: If the Password parameter has an asterisk as its value and is loaded to the device, it has no affect on the device's configuration (i.e., the existing value of the Password parameter is retained). Version February 2010
26 Mediant 2000 & Mediant SRTP Option without SDP Capability Negotiation: Mediant 2000 Mediant 3000 Mediant 3000 In previous releases, the device supported two security modes (configured by the parameter MediaSecurityBehaviour): Mandatory mode: The device initiates encrypted calls, but if negotiation of the cipher suite fails, the call is terminated. Incoming calls that do not include encryption information are rejected. Preferable mode: The device initiates encrypted calls. If negotiation of the cipher suite fails, an un-encrypted call is established. Incoming calls that do not include encryption information are accepted (default). In this mode, the device initiates SDP with two media lines (m=) - one for RTP and one for SRTP. In this release, the device supports an additional mode, "Preferable - Single Media", also configured by the existing MediaSecurityBehaviour parameter. This mode is the same as the Preferable mode, except for the following differences: Instead of two "m=" lines in the suggested SDP, it uses only a single m= line. Instead of a media line with RTP/SAVP, it uses RTP/AVP. In addition, in this mode, if the remote SIP UA does not support SRTP, it ignores the crypto lines. An example of an SDP with one "m=" line and crypto, v=0 o=audiocodesgw IN IP s=phone-call c=in IP t=0 0 m=audio 6000 RTP/AVP a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:70 EG711A/8000 a=rtpmap:96 telephone-event/8000 a=fmtp: a=ptime:30 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9lxqem1/dgtln2ehp46jfuxrpgpxdwpu/bmsvu9l 2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:5orhfgojn8onjoorisznrcpegbhmv5d3ji9wqbp4 2^31 This feature can also be assigned to an IP Profile. Note that for this feature to be functional, the EnableMediaSecurity parameter must be set to 1. Relevant parameters: MediaSecurityBehaviour; IPProfile. SIP Release Notes 26 Document #: LTRT-69017
27 SIP Release Notes 1. What's New in Release Retain Active Call's SRTP Key after Blade Switchover: Mediant 2000 Mediant 3000 Mediant 3000 The device now retains (uses) the same Secure Real-time Transport Protocol (SRTP) key of an active call after a blade switchover has occurred. In previous releases, the device changed SRTP keys in its offer or answer if a Re-INVITE was sent or received for an active call after switchover to the redundant blade. This sometimes resulted in one-way voice. 35. Sending Re-INVITE with New SRTP Key upon Receipt of 181 Response: Mediant 2000 Mediant 3000 Mediant 3000 The device now can be configured to send a Re-INVITE with a new SRTP key upon receipt of a SIP 181 response ("call is being forwarded"). This is in accordance with the UCR 2008 standard. If the device sends an INVITE with SDP and receives a 181 response, it changes the SRTP key by sending a Re-INVITE with a new SRTP key in its SDP. Relevant parameter: EnableRekeyAfter Maximum SAS Registrations: Mediant 2000 Mediant 3000 Mediant 3000 The device now rejects additional SAS registrations for endpoints if the maximum number of registrations has been reached. For fully populated chassis, the device will not register more than 500 endpoints (i.e., the 501 endpoint registration will be rejected). For depopulated chassis configuration, the maximum SAS endpoint registrations is 2, Interworking SAS behind NAT: Mediant 2000 Mediant 3000 Mediant 3000 Since SAS is implemented as a standard proxy, it s default behavior while relaying requests is as follows: Adds a new SIP Via header (with the SAS IP address) as the top-most Via header. Does not modify the original SIP Contact header. This default and standard proxy result in the top-most Via header and the Contact header to point to different hosts. However, some SBC s (e.g., ACME) require that incoming requests must point to the same host in the top-most Via header and the Contact header. For interoperability support with such an SBC, the device can now operate in a new mode that changes the Contact header so that it points to the SAS host, and therefore, the top-most Via header and the Contact header point to the same host. Version February 2010
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