Session Initiation Protocol (SIP) for VoIP

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Session Initiation Protocol (SIP) for VoIP

  1. 1. Session Initiation Protocol (SIP) for VoIP Document Update Alert This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available. If you are using Cisco IOS Release 12.3 or higher, refer to the following documentation in the Cisco IOS Voice Configuration Library, Release 12.3: • Cisco IOS SIP Configuration Guide If you are using Cisco IOS Release 12.2 or higher, refer to the following chapter in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2: • Configuring Session Initiation Protocol for Voice over IP Feature History Release Modification 12.1(1)T SIP was introduced on Cisco Access platforms. 12.1(3)T SIP Enhancements were implemented on Cisco 2600 series and Cisco 3600 series routers. 12.1(3)XI The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was introduced and implemented on Cisco 2600 series, Cisco 3600 series routers, and the Cisco AS5300 universal access server. 12.1(5)T The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was integrated into Cisco IOS Release 12.1(5)T. 12.2(2)T SIP Enhancements were integrated into Cisco IOS release 12.2(2)T and implemented on the Cisco AS5400 universal gateway. The SIP User Agent MIB feature was introduced and implemented on the Cisco 2600 series and Cisco 3600 series routers. The SIP Diversion Header Implementation for Redirecting Number feature was introduced and implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300 universal access servers. 12.2(2)XA SIP and SIP Enhancements were integrated in Cisco IOS Release 12.2(2)XA and implemented on the Cisco AS5400 and AS5350 universal gateways. Cisco IOS Release 12.2(8)T and 12.2(11)T 1
  2. 2. Session Initiation Protocol (SIP) for VoIP 12.2(2)XB The SIP Gateway Support for Bind Command, SIP Gateway Support of RSVP and TEL URL, SIP INVITE Request with Malformed Via Header, Configurable PSTN Cause Code to SIP Response Mapping, RFC2782 Compliance for DNS SRV, SIP T.38 Fax Relay, and Call Transfer Capabilities Using the Refer Method features were introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300 universal access server, Cisco AS5350 and Cisco AS5400 universal gateways. 12.2(2)XB1 The SIP T.38 Fax Relay feature was implemented on the Cisco AS5300 universal access server, Cisco AS5350, and AS5400 universal gateways. SIP, SIP Enhancements, and SIP Gateway Support of RSVP and TEL URL features were implemented on the Cisco AS5850 universal gateway. 12.2(2)XB2 The SIP Gateway Support for Bind Command, Configurable PSTN Cause Code to SIP Response Mapping, Call Transfer Capabilities Using the Refer Method, and SIP T.38 Fax Relay features were implemented on the Cisco AS5850 universal gateway. 12.2(4)XM The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was implemented on Cisco 1700 series routers. 12.2(8)T SIP, and the following SIP features were implemented on the Cisco 7200 series routers: SIP Enhancements, DTMF Relay for SIP Calls Using Named Telephone Events, SIP User Agent MIB, ISDN Progress Indicator Support for SIP Using 183 Session Progress, SIP Diversion Header Implementation for Redirecting Number, SIP Gateway Support of RSVP and TEL URL, SIP Intra-gateway Hairpinning, SIP INVITE Request with Malformed Via Header, Configurable PSTN Cause Code to SIP Response Mapping, RFC 2782 Compliance for DNS SRV, Call Transfer Capabilities Using Refer, and SIP T.38 Fax Relay features were integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release. 12.2(11)T This feature was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway. This document describes the Session Initiation Protocol (SIP) for VoIP on Cisco 7200 series routers in Cisco IOS Release 12.2(8)T and contains the following sections: • Feature Overview, page 3 • Supported Platforms, page 14 • Supported Standards, MIBs, and RFCs, page 15 • Prerequisites, page 16 • Configuration Tasks, page 16 • Configuration Examples, page 24 • Command Reference, page 32 • Glossary, page 94 Cisco IOS Release 12.2(8)T and 12.2(11)T 2
  3. 3. Session Initiation Protocol (SIP) for VoIP Feature Overview Feature Overview Session Initiation Protocol (SIP) Voice over Internet Protocol (VoIP) currently implements ITU’s H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. The Cisco SIP functionality equips Cisco routers to signal the setup of voice and multimedia calls over IP networks. SIP provides an alternative to H.323 within the VoIP internetworking software. The SIP feature also provides nonproprietary advantages in the areas of: • Protocol extensibility • System scalability • Personal mobility services • Interoperability with different vendors The SIP feature includes the following functionality: • Configurable in-band alerting • Ability to specify the maximum number of SIP redirects • Ability to specify SIP or H.323 on a dial-peer basis • Support for both User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) transport layers for SIP messages • Powerful debugging support • Support for Domain Name System Server (DNS SRV) records for resolving SIP server host names • Configurable SIP message timers and retries SIP Enhancements Beginning in Cisco IOS Release 12.1(3)T, the following enhancements to SIP were introduced: • Configurable SIP message timers and retries • Interoperability with unified call services (UCS) • Support for a variety of signaling protocols, including ISDN, PRI, and CAS • Support for a variety of interfaces, including – Analog interfaces: FXS/FXO/E&M analog interfaces – Digital interfaces: T1 CAS and E1 CAS • Support for SIP redirection messages and interaction with SIP proxies. The gateway can redirect an unanswered call to another SIP gateway or SIP-enabled IP phone. In addition, the gateway supports proxy-routed calls. • Interoperability with DNS servers including support for DNS SRV and “A” records to look up SIP URLs • Support for SIP over TCP and UDP network protocols • Support for Routing Table Protocol/RTP Control Protocol (RTP/RTCP) for media transport in VoIP networks • Support for the following codecs (see Table 1): Cisco IOS Release 12.2(8)T and 12.2(11)T 3
  4. 4. Session Initiation Protocol (SIP) for VoIP Feature Overview Table 1 SIP-Supported Codecs Codec SDP G711ulaw 0 G711alaw 8 G723r63 4 G726r16 2 G728 15 G729r8 18 • Support for Record-Route headers • Support for IP Quality of Service (QoS) and IP precedence • Support for IP Security (IPSec) for SIP signalling messages • Authentication, Authorization, and Accounting (AAA) support. For accounting, the gateway device generates call data record (CDR) accounting records for export. For authentication, the SIP Gateway sends validate requests to the AAA server. For authorization, the existing access lists are used. • Support for configurable expiration time for SIP INVITEs and maximum number of proxies or redirect servers that can forward a SIP request • Expanded support for the mapping of Public Switched Telephone Network (PSTN) cause codes to SIP events • Ability to hide the calling party’s identity based on the setting of the ISDN presentation indicator For more information, see the Configuring SIP for VoIP part in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2. Call Transfer Capabilities Using the Refer Method The Refer method provides call transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS Session Initiation Protocol (SIP) gateways. Call transfer allows a wide variety of decentralized multiparty call operations. These decentralized call operations form the basis for third-party call control, and thus are important features for Voice over IP (VoIP) and SIP. Call transfer is also critical for conference calling, where calls can transition smoothly between multiple point-to-point links and IP-level multicasting. For more information, see the document Call Transfer Capabilities Using the Refer Method. Configurable PSTN Cause Code to SIP Response Mapping This feature allows customization of the standard RFC 2543 mappings between the Session Initiation Protocol (SIP) network and the Public Switched Telephone Network (PSTN). For calls to be established between a SIP network and a PSTN, the two networks must be able to interoperate. One aspect of their interoperation is the mapping of PSTN cause codes, which indicate reasons for PSTN call failure or completion, to SIP status codes or events. The opposite is also true: SIP status codes or events are mapped to PSTN cause codes. Event mapping tables found in this document show the standard or default mappings between SIP and PSTN. Cisco IOS Release 12.2(8)T and 12.2(11)T 4
  5. 5. Session Initiation Protocol (SIP) for VoIP Feature Overview However, you may want to customize the SIP user agent software to override the default mappings between the SIP network and the PSTN. The Configurable PSTN Cause Code to SIP Response Mapping feature allows you to configure specific map settings between the PSTN and SIP networks. Thus, any SIP status code can be mapped to any PSTN cause code, or vice versa. When set, these settings can be stored in the NVRAM and are restored automatically on bootup. For more information about this feature, including configuration tasks and examples, see the document Configurable PSTN Cause Code to SIP Response Mapping. DTMF Relay for SIP Calls Using Named Telephone Events The DTMF Relay for SIP calls Using Named Telephone Events (NTE) feature adds support for relaying DTMF tones and hookflash events in SIP on Cisco VoIP gateways. Note The DTMF Relay for SIP Calls Using Named Telephone Events feature is implemented for SIP only. Using NTE to relay dual tone multifrequency (DTMF) tones provides a standardized means of transporting DTMF tones in RTP packets according to section 3 of RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, developed by the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group. RFC 2833 defines formats of NTE RTP packets used to transport DTMF digits, hookflash, and other telephony events between two peer endpoints. DTMF tones are generated when a button on a touch-tone phone is pressed. When the tone is generated, it is compressed, transported to the other party, and decompressed. If a low-bandwidth codec, such as a G.729 or G.723 is used without a DTMF relay method, the tone may be distorted during compression and decompression. The DTMF Relay for SIP Calls Using NTE feature adds SIP functionality. SIP IP phones currently do not have the capability to generate DTMF tones. Currently, DTMF tones are transferred using Cisco Proprietary RTP or transparently in band. The DTMF Relay using NTE feature allows SIP phones calling voice mail or other interactive voice response (IVR) systems to relay DTMF tones. Additionally, this feature prevents distortion of DTMF tones if the RTP session uses a low bit-rate codec, because tones are passed in NTE packets and are not compressed using the default codec. With the DTMF Relay Using NTE feature, the endpoints can perform per-call negotiation of the DTMF relay method. During call setup, the calling and called parties negotiate to choose the DTMF relay mode. They also negotiate to determine the payload type value for the NTE RTP packets. In a SIP call, the gateway forms a session description protocol (SDP) message that indicates: • If NTP will be used • Which events will be sent using NTE • NTE payload type value The DTMF Relay Using NTE feature also provides hookflash support using in-band and out-of-band modes. In in-band mode, the gateway relays the hookflash without notifying the application, and the default session application and any IVR scripts do not receive the hookflash. In out-of-band mode, the gateway reports the hookflash to the application and the application can relay the hookflash to the next call leg. Note In addition, the DTMF Relay for SIP Calls Using NTE feature does not support hookflash generation for advanced features such as call waiting and conferencing. For more information, see the document DTMF Relay Using Named Telephone Event. Cisco IOS Release 12.2(8)T and 12.2(11)T 5
  6. 6. Session Initiation Protocol (SIP) for VoIP Feature Overview ISDN Progress Indicator Support for SIP Using 183 Session Progress This feature provides support for handling inband treatments, such as call progress tones and announcements, when SIP is the session protocol for establishing call connections. The feature ensures the correct establishment of the media stream through the SIP network to allow the successful transport of in-band treatments, which might ingress from a PSTN node on a SIP gateway or egress to a PSTN node. The feature also allows VoIP calls using SIP to provide inband call treatment such as ringback tones, announcements when interworking with ISDN and channel associated signaling (CAS) PSTN networks. SIP 183 Session Progress messages facilitate better call treatment for SIP VoIP calls when interworking with PSTN networks. The introduction of the 183 Session Progress message allows a called user agent to suppress local alerting from the calling user agent, and to play a tone or announcement during a preliminary call session, before the full SIP session is set up. This functionality enables the calling party to be notified of the status of the call without being charged for the preliminary portion of the call. A new Session header in the 183 Session Progress message controls whether or not the called user agent plays a tone or announcement for the calling party. The 183 Session Progress message is supported by default and does not require any special configuration. RFC 2782 Compliance for DNS SRV SIP on Cisco’s VoIP gateways uses DNS SRV query to determine the IP address of the SIP Proxy or the Redirect Server. The query string generated has a prefix in the form of “protocol.transport.” and is attached to the Fully Qualified Domain Name (FQDN) of the next hop SIP server. This prefix style from RFC 2052 has always been available; however, with this release a second style is also available. The second style is in compliance with RFC 2782, and prepends the protocol label with an underscore “_”; as in “_protocol._transport.”. The addition of the underscore reduces the risk of the same name being used for unrelated purposes. Use the srv version command to configure the DNS SRV feature. For more information, see the document SIP Gateway Support of RSVP and TEL URL. SIP Diversion Header Implementation for Redirecting Number The SIP Diversion Header Implementation for Redirecting Number feature provides support for a new SIP header field; Call Control (CC)-Diversion. The CC-Diversion header field enables the SIP gateway to pass call control redirecting information during the call setup. Call control redirection is the redirection of a call based on a subscriber service such as call forwarding. Call redirection information is information typically used for Unified Messaging and voice mail services to identify the recipient of a message. Call control rediversion information can also be used to support applications such as automatic call distribution, and enhanced telephony features such as Do Not Disturb and Caller ID. If generated by the SIP gateway during call process, the CC-Diversion header field is based on the contents of the Redirecting Number Information Element (IE) in the ISDN Setup message. In addition, information such as the reason the call was redirected is included in the CC-Diversion header field. For more information, see the document SIP Diversion Header Implementation for Redirecting Number. SIP Gateway Support for Bind Command Currently, Session Initial Protocol (SIP) signaling and media paths use an IP address that is provided by the IP layer as the source address. However, with the addition of the bind command, you can now configure the source IP address of signaling packets, or both signaling and media packets. In previous releases of Cisco IOS software, the source address of a packet going out of the gateway was never deterministic. That is, the session protocols and VoIP layers always depended on the IP layer to give the best local address. The best local address was then used as the source address (the address Cisco IOS Release 12.2(8)T and 12.2(11)T 6
  7. 7. Session Initiation Protocol (SIP) for VoIP Feature Overview showing where the SIP request came from) for signaling and media packets. Using this nondeterministic address occasionally caused confusion for firewall applications, as a firewall could not be configured with an exact address and would take action on several different source address packets. However, the bind interface command allows you to configure the source IP address of signaling and media packets to a specific interface’s IP address. Thus, the address that goes out on the packet is bound to the IP address of the interface specified with the bind command. Packets that are not destined to the bound address are discarded. When you do not want to specify a bind address, or if the interface is down, the IP layer still provides the best local address. The bind command performs different functions based on the state of the interface. For more information, see the document SIP Gateway Support for the Bind Command. SIP Gateway Support of RSVP and TEL URL The SIP Gateway Support of RSVP and TEL URL feature provides the following SIP enhancements: • RSVP • Telephone URL format in SIP messages • Interaction with forking proxies • SIP intra-gateway hairpinning • Reliability of SIP provisional responses • Configurable screening indicator • RFC 2782 Compliance (style of DNS SRV queries) For more information, see the document SIP Gateway Support of RSVP and TEL URL. SIP Intra-Gateway Hairpinning SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This can be a Public Switched Telephone Network (PSTN) call routed into the IP network and back out to the PSTN over the same gateway, as shown below: Gateway PSTN IP network call id - x 37698 call id - x Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway: Gateway Line 1 IP network call id - y 37699 Line 2 call id - y With SIP hairpinning, unique gateways for ingress and egress are no longer necessary. For more information about the SIP Intra-Gateway Hairpinning feature, including configuration tasks and examples, see the document SIP Gateway Support of RSVP and TEL URL. Cisco IOS Release 12.2(8)T and 12.2(11)T 7
  8. 8. Session Initiation Protocol (SIP) for VoIP Feature Overview SIP INVITE Request with Malformed Via Header A SIP INVITE requests that a user or service participate in a session. Each INVITE contains a Via header that indicates the transport path taken by the request so far, and where to send a response. In the past, when an INVITE contained a malformed Via header, the gateway would print a debug message and discard the INVITE without incrementing a counter. However, the printed debug message was often inadequate, and it was difficult to detect that messages were being discarded. The SIP INVITE Request with Malformed Via Header feature provides a response to the malformed request. A counter, Client Error: Bad Request, increments when a response is sent for a malformed Via field. Bad Request is a class 400 response and includes the explanation Malformed Via Field. The response is sent to the source IP address (the IP address where the SIP request originated) at User Datagram Protocol (UDP) port 5060. Note This feature applies to messages arriving on UDP, because the Via header is not used to respond to messages arriving on TCP. For more information about this feature, see the document SIP INVITE Request with Malformed Via Header. SIP T.38 Fax Relay The SIP T.38 Fax Relay feature adds standards-based fax support to SIP and conforms to ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks. The ITU-T standard specifies real-time transmission of faxes between two regular fax terminals over an IP network. Much like a voice call, SIP T.38 Fax Relay requires call establishment, data transmission, and release signaling. The following figure shows the basic setup of SIP T.38 Fax Relay: IP network T.38 path 62318 SIP SIP originating gateway terminating gateway For more information, including configuration tasks and examples, see the document SIP T.38 Fax Relay. SIP User Agent MIB The SIP User Agent MIB addresses the need for SIP-specific gateway information to be made available by Simple Network Management Protocol (SNMP). The implementation of this capability is based upon the current IETF draft “draft-ietf-sip-mib-01.txt”. The implementation of the SIP MIB in the Cisco SIP gateway supports configuration objects related to SIP such as the configured SIP server, SIP timers, and number of retry attempts allowed for requests and responses. The SIP MIB also supports SIP-specific statistical information objects. This includes information on numbers of provisional responses, success responses, redirection responses, client error responses, server error responses, and global error responses. In addition, the SIP MIB includes information regarding SIP Requests initiated and received as well as information about retries associated with each SIP Request type. Cisco IOS Release 12.2(8)T and 12.2(11)T 8
  9. 9. Session Initiation Protocol (SIP) for VoIP Feature Overview Benefits Session Initiation Protocol The SIP feature meets the needs of service providers that use SIP on the gateways of their VoIP network to: • Enable Cisco voice-enabled platforms to provide RFC 2543-compliant user-agent client gateways • Support codecs capable of Carrier-class voice quality Although SIP is simpler than H.323, SIP provides similar capabilities in: • System scalability • End-to-end solutions • High-density voice gateways SIP Enhancements The SIP feature enhancements enable SIP gateways to: • Enable Cisco voice-enabled platforms to provide RFC 2543-compliant user-agent client gateways • Support proxy-routed calls • Redirect an unanswered call to another SIP gateway or SIP-enabled IP phone • Allow end users to place calls on hold • Hide the calling party’s identity based on the setting of the ISDN presentation indicator Call Transfer Capabilities Using the Refer Method • SIP Call Transfer Using the Refer Method supports attended transfer and blind transfer in accordance with emerging SIP standards. Configurable PSTN Cause Code to SIP Response Mapping • The Configurable PSTN Cause Code to SIP Response Mapping feature offers control and flexibility. By using command-line interface commands, you can easily customize the default or standard mappings that are currently available between PSTN and SIP networks. This allows for flexibility when setting up deployment sites. DTMF Relay for SIP Calls Using Named Telephone Events • DTMF relay support for SIP • Hookflash relay support for SIP • Simultaneous support with Cisco Proprietary RTP (used for modem passthrough and modem relay) • Provisioning of RTP payload type values • Per-call negotiation of relay method and payload type values • More accurate tone delivery • Interoperability with SIP applications from other vendors Cisco IOS Release 12.2(8)T and 12.2(11)T 9
  10. 10. Session Initiation Protocol (SIP) for VoIP Feature Overview ISDN Progress Indicator Support for SIP Using 183 Session Progress • Ensures that in-band treatments initiated in the PSTN are successfully transported through the SIP network • Allows for internetworking of features between the PSTN and the SIP network so that the correct inband feedback is provided to the feature user RFC 2782 Compliance for DNS SRV • Compliance with RFC 2782 brings DNS compatibility. RFC 2782 updates RFC 2052 by prepending the protocol label with an underscore “_”. This change reduces the risk of the same name being used for unrelated purposes. However, backward compatibility is available, allowing newer versions of IOS software to work with older networks that only support RFC 2052. • Currently you must know the exact address of a server to contact it. SRV records enable administrators to use several servers to provide the same service within a single domain. SRV Resource Records (RRs) allow administrators to define primary and backup servers and move services from host-to-host without affecting service. SIP Diversion Header Implementation for Redirecting Number • Provides support for the Call Control (CC)-Diversion SIP header field • Enables the SIP gateway to pass call control redirecting information during the call setup • Redirection of a call based on a subscriber service such as call forwarding • Unified Messaging and voice mail services to identify the recipient of a message • Support of applications such as automatic call distribution, and enhanced telephony features such as Do Not Disturb and Caller ID SIP Gateway Support for the Bind Command • With the bind command, SIP signaling and media paths can advertise the same source IP address on the gateway for certain applications, even if the paths used different addresses to reach the source. This eliminates confusion for firewall applications that, prior to the use of binding, may have taken action on several different source address packets. SIP Gateway Support of RSVP and TEL URL • SIP Gateway Support of RSVP and TEL URL enables QoS, ensuring certain bandwidth reservations for specific calls. The bandwidth reservation can be best-effort, in which case the call is completed even if the reservation is not supported by both sides or cannot be established. Or the bandwidth reservation can be required, and the call is not set up if the bandwidth reservation is not performed successfully. • With the reliable provisional response features, you can ensure that media information is exchanged and resource reservation takes place before connecting a call. • Forked call responses to Cisco IOS gateways are now supported. Call forking enables the terminating gateway to handle multiple requests and the originating gateway to handle multiple provisional responses for the same call. Call forking is required for the deployment of the find me/follow me type of services. • Gateways now accept TEL calls sent through the Internet, which provides interoperability with other equipment that uses TEL URL. The TEL URL feature also gives service providers a way to differentiate services based on the type of call, allowing for deployment of specific services. Cisco IOS Release 12.2(8)T and 12.2(11)T 10
  11. 11. Session Initiation Protocol (SIP) for VoIP Feature Overview SIP Intra-Gateway Hairpinning • Hairpinning enables the same gateway to originate and terminate a call. It works independently of, but enhances, call forking. It also enables call control features to function when it is required that an incoming PSTN call is routed back out to the PSTN on the same Cisco gateway. SIP INVITE Request with Malformed Via Header • Incrementing a counter and sending a response, rather than simply discarding the INVITE, if it contains a malformed Via header. The counter provides a useful and immediate indication that an INVITE has been discarded, and the response allows the result to be propagated back to the sender. SIP T.38 Fax Relay • Cisco furthers its commitment to open standards and to the success of its customers by supporting standards such as ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks and T-38 Annex-D. • T.38 Fax Relay over packet networks has become a popular way to bypass tolls associated with sending faxes. SIP T.38 Fax Relay provides standards-based toll bypass for both fax and voice calls. Toll bypass capabilities can result in cost savings to end users of packet telephony networks. • A Cisco originating gateway (OGW) that has T.38 support automatically enters T.38 mode if it receives a T.38 INVITE, even if it is configured for the Cisco proprietary Fax Relay. This choice of fax protocols provides an extremely reliable fax transfer mechanism. • Currently, SIP uses the Cisco proprietary Fax Relay solution. However, Cisco Fax Relay is sometimes not an ideal solution for enterprise and service provider customers who have implemented a multivendor network. Because the T.38 Fax Relay protocol is standards-based, Cisco gateways and gatekeepers can operate with third-party T.38-enabled gateways and gatekeepers in a multivendor network where real-time Fax Relay capabilities are required. • T.38 Fax Relay is already implemented in Cisco gateways that support H.323 and Media Gateway Control Protocol (MGCP). The addition of T.38 for SIP strengthens SIPs position as a low-cost standards-based infrastructure, and increases its viability as the protocol of choice for next-generation IP networks. SIP User Agent MIB • The SIP User Agent MIB provides SIP-specific information via SNMP—this information allows customers to have SIP-specific information available to evaluate the performance of gateways in conjunction with their SIP networks. Restrictions SIP Ensure that your access platform has 16 MB Flash memory and 64 MB DRAM memory minimum, and that I/O memory is set to either 8 or 16 MB. SIP Enhancements • The SIP Gateway does not support codecs other than those listed in Table 1 on page 4. – If on the originating gateway, an appropriate SIP debug trace is presented, indicating the failure to originate the SIP call leg. – If on the terminating gateway, an appropriate SIP response (4xx) with a warning indicating incompatible media types is sent. • The SIP Gateway requires each INVITE to include a Session Description Protocol (SDP) header. Cisco IOS Release 12.2(8)T and 12.2(11)T 11
  12. 12. Session Initiation Protocol (SIP) for VoIP Feature Overview • The contents of the Session Description Protocol (SDP) header cannot change between the 180 Ringing message and the 200 OK message. • SIP requires that all times be sent in Greenwich Mean Time (GMT). The INVITE is sent with GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify the GMT time. • The Enhancements to SIP for VoIP feature supports plain old telephone service (POTS) to POTS hairpinning (which means the call comes in one voice port and is router out another voice port). It also supports POTS to IP call legs and IP to POTS call legs. However, it does not support IP to IP hairpinning. This means the SIP Gateway cannot take an inbound SIP call and reroute it back to another SIP device using the VoIP dial peers. • SIP requires that all times be sent in Greenwich Mean Time (GMT). The INVITE is sent with GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify the GMT. • VoIP dial peers allow a user to configure the bytes parameter associated with a codec. However, Cisco SIP gateways currently do not present or respond to this parameter. Currently, the a=ptime parameter is not sent or recognized in the SDP body of a SIP message. • With call transfer, the Requested-By header identifies the party initiating the transfer. The Requested-By header is included in the INVITE request that is sent to the transferred-to party only if a Requested-By header was also included in the Bye request. • With call transfer, the Also header identifies the transferred-to party. To invoke a transfer, the user portion of the Also header must be defined explicitly or with wildcards as a destination pattern on a VoIP dial peer. The transferred call is routed using the session target parameter on the dial peer instead of the host portion of the Also header. Therefore, the Also header can contain user@host but the host portion is ignored for call routing purposes. • The grammar for the Also and Requested-By headers is not fully supported. Only the name-addr header is supported. This implies that the crypto-param, which might be present in the Bye request, will not be populated in the ensuing INVITE to the transferred-to party. • Cisco SIP Gateways do not support the user=np-queried parameter in a Request URI. • If a Cisco SIP Gateway receives an ISDN Progress message, it generates a 183 Session Progress message. If the gateway receives an ISDN ALERT, it generates a 180 Ringing message. Call Transfer Capabilities Using the Refer Method • Although SIP IOS gateways currently support SIP URLs and TEL URLs, the Refer-To header must be in SIP URL format to be valid. The TEL URL format cannot be used, because it does not provide a host portion, and without one, the triggered Invite request cannot be routed. • Only three overloaded headers in the Refer-To header are accepted: Accept-Contact, Proxy-Authorization, and Replaces. All other headers present in the Refer-To are ignored. • The Refer-To and Contact headers are required in the Refer request. The absence of either header results in a 4xx class response to the Refer request. Also, the Refer request must contain exactly one Refer-To header. Multiple Refer-To headers result in a 4xx class response. • The Referred-By header is required in a Refer request. The absence of this header results in a 4xx class response to the Refer request. Also, the Refer request must contain exactly one Referred-By header. Multiple Referred-By headers result in a 4xx class response. • As with the Bye and Also call transfer methods, the dial peers must be configured for correct functioning of the Refer method. Cisco IOS Release 12.2(8)T and 12.2(11)T 12
  13. 13. Session Initiation Protocol (SIP) for VoIP Feature Overview DTMF Relay for SIP Calls Using Named Telephone Events • The DTMF Relay for SIP calls Using Named Telephone Events feature is only available on Cisco VoIP gateways using SIP. The DTMF Relay for SIP Calls Using NTE feature does not support hookflash generation for advanced features such as call waiting and conferencing. SIP Gateway Support of RSVP and TEL URL • Support for interaction with forking proxies applied only to gateways acting as a user agent client (UAC) is not supported. It does not apply when the gateway acts as a user agent server (UAS). In that case, the proxy forks multiple INVITES with the same call ID to the same gateway but with different request URLs. • Forking functionality sets up RSVP for each transaction only if the dial peers are configured for QoS. If not, the calls proceed as best-effort. Bandwidth reservation (QoS) is not supported for Session Description Protocol (SDP changes between 183 Session Progress/180 Alerting and 200 OK responses). • Bandwidth reservation (QoS) is not attempted if the desired QoS level is set to the default of best-effort. The desired QoS for the associated dial peer must be set to controlled-load or guaranteed-delay. • Distributed Call Signaling (DCS) headers and extensions are not supported. SIP INVITE Request with Malformed Via Header • Distributed Call Signaling (DCS) headers and extensions are not supported. SIP T.38 Fax Relay • For SIP T.38 Fax Relay only UDP is supported for the transport layer. • If SIP T.38 Fax Relay is not supported by both gateways, the T.38 negotiation fails and the call reverts back to an audio codec. • T.38 Fax Relay requires 64 Kbps, the same amount of bandwidth as a voice call with the G.711 codec. • Calling tones (CNG) are optional, and are not used to initiate a switch to T.38 mode. Instead, called terminal identification tones (CED) or preamble flags are used. • This feature does not rely on named signaling events (NSE) to signal a switch to T.38 mode. Standard RFC 2543 and RFC 2327 SIP and SDP signaling are used instead. Note The transport protocols specified in the ITU-T Recommendation for T.38 are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). However, only UDP is supported for Cisco IOS Release 12.2(2)XB. For further information on T.38 protocol, refer to the ITU-T Recommendations. Related Features and Technologies • Cisco Fax Relay • Cisco IP Phones • Cisco QoS • Cisco RSVP • Cisco SIP Proxy Server Cisco IOS Release 12.2(8)T and 12.2(11)T 13
  14. 14. Session Initiation Protocol (SIP) for VoIP Supported Platforms • Cisco TCL/IVR Version 2.0 • Cisco VoIP Related Documents The following documents contain information related to Cisco SIP functionality: • Cisco IOS IP Configuration Guide, Release 12.2 • Cisco IOS IP Command Reference, Volume 1 of 3: Addressing and Services, Release 12.2 • Cisco IOS IP Command Reference, Volume 2 of 3: Routing Protocols, Release 12.2 • Cisco IOS IP Command Reference, Volume 3 of 3: Multicast, Release 12.2 • Cisco IOS Quality of Service Solutions Command Reference, Release 12.2 • Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2 • Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 • Cisco IP Telephony Network Design Guide • Configuring Session Initiation Protocol for Voice over IP • Dial Peer Enhancements • Service Provider Features for Voice over IP, Release 12.0(3)T • Session Initiation Protocol Call Flows • Session Initiation Protocol Gateway Call Flows • Session Initiation Protocol Gateway Call Flows and Compliance Information • SIP Call Flows, Release 12.2(4)T • SIP Diversion Header Implementation for Redirecting Number • SIP Gateway Support of RSVP and TEL URL, Release 12.2(2)XB • TCL IVR API Version 2.0 Programmer's Guide • VoIP Call Admission Control Using RSVP chapter in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 • Voice over IP for the Cisco 2600/3600 Series Supported Platforms • Cisco 2691 • Cisco 3631 • Cisco 3725 • Cisco 3745 • Cisco 7200 series • Cisco AS5850 Cisco IOS Release 12.2(8)T and 12.2(11)T 14
  15. 15. Session Initiation Protocol (SIP) for VoIP Supported Standards, MIBs, and RFCs Determining Platform Support Through Cisco Feature Navigator Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature. Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common. To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions at http://www.cisco.com/register. Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL: http://www.cisco.com/go/fn Supported Standards, MIBs, and RFCs Standards • ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks • ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks, Amendment 1 • ITU-T, T.38, Annex-D MIBs • CISCO-SIP-UA-MIB To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL: http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml RFCs • RFC 1890, RTP Profile for Audio and Video Conferences with Minimal Control • RFC 2327, SIP/SDP Signaling • RFC 2543, SIP: Session Initiation Protocol • RFC 2728, A DNS RR for Specifying the Location of Services (DNS SRV) • RFC 2806, URLs for Telephone Calls • RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals Cisco IOS Release 12.2(8)T and 12.2(11)T 15
  16. 16. Session Initiation Protocol (SIP) for VoIP Prerequisites Prerequisites General SIP Prerequisites • Your gateway must have voice functionality that is configurable for either SIP. • Establish a working IP network. • Configure VoIP. • Ensure that your Cisco 2600 series, Cisco 3600 series, or Cisco 7200 series router has 16-MB Flash memory and 64-MB DRAM memory, minimum. SIP Gateway Support for Bind Command • Set the bind address prior to using the bind command. Call Transfer Capabilities Using the Refer Method • Configure the SIP dial peers for call transfer. As with the Bye and Also call transfer methods, the dial peers must be configured for correct functioning of the Refer method. See the document Call Transfer Capabilities Using the Refer Method for complete configuration steps. Configuration Tasks SIP See the following sections for configuration tasks for basic SIP functions. Each task in the list is identified as either required or optional. • Configuring the SIP User Agent (UA) (required) • Changing the Configuration of the SIP User Agent (UA) (optional) • Configuring SIP Support for VoIP Dial Peers (optional) • Configuring a POTS Dial Peer (optional) • Configuring SIP Call Transfer for a POTS Dial Peer (optional) • Configuring SIP Call Transfer for a VoIP Dial Peer (optional) • Configuring Phone Number Translation Rules (required) • Verifying the SIP Feature Configuration (optional) For more information on SIP configuration, including call flows, refer to the Session Initiation Protocol Call Flows document. Call Transfer Capabilities Using the Refer Method For configuration tasks for this feature, see the document Call Transfer Capabilities Using the Refer Method. Configurable PSTN Cause Code to SIP Response Mapping For configuration tasks for this feature, see the document Configurable PSTN Cause Code to SIP Response Mapping. Cisco IOS Release 12.2(8)T and 12.2(11)T 16
  17. 17. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events For configuration tasks for this feature, see the document Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events. ISDN Progress Indicator Support for SIP Using 183 Session Progress There are no configuration tasks for this feature. RFC 2782 Compliance for DNS SRV For configuration tasks for this feature, see the document SIP Gateway Support of RSVP and TEL URL. SIP Diversion Header Implementation for Redirecting Number For configuration tasks for this feature, see the document SIP Diversion Header Implementation for Redirecting Number. SIP Gateway Support for Bind Command For configuration tasks for this feature, see the document SIP Gateway Support for Bind Command. SIP Gateway Support of RSVP and TEL URL For configuration tasks for this feature, see the document SIP Gateway Support of RSVP and TEL URL. SIP Intra-Gateway Hairpinning There are no configuration tasks for this feature. SIP INVITE Request with Malformed Via Header There are no configuration tasks for this feature. SIP T.38 Fax Relay For configuration tasks for this feature, see the document SIP T.38 Fax Relay. SIP User Agent MIB There are no configuration tasks for this feature. Cisco IOS Release 12.2(8)T and 12.2(11)T 17
  18. 18. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Configuring the SIP User Agent (UA) A terminating gateway that is not configured as an SIP user agent cannot receive incoming SIP calls. The transport command opens the SIP listener port (5060) to receive SIP (a SIP user agent is configured to listen by default). To configure the terminating gateway, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# sip-ua Enters SIP user-agent mode to configure SIP UA-related commands. Step 2 Router(config-sip-ua)# transport Configures the SIP user agent (sip-ua) for SIP signaling {udp | tcp} messages. The default value is udp. • udp—Configures the SIP user agent to receive SIP messages on UDP port 5060. • tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060. Step 3 Router(config-sip-ua)# sip-server Enters the IP address of the SIP server interface. ipv4:ip-address Step 4 Router(config-sip-ua)# timers trying Sets time to wait for a response. number • number—Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500. Step 5 Router(config-sip-ua)# retry invite Configures the SIP signaling timers for retry attempts. number • number—Number of INVITE retries: 1 through 10 are valid inputs; default = 6. Changing the Configuration of the SIP User Agent (UA) It is not necessary to configure a SIP UA in order to place a call. A SIP UA is configured to listen by default. However, if you want to adjust any of the settings, you can do so using the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# sip-ua Enters SIP user agent (sip-ua) mode to configure SIP UA-related commands. Step 2 Router(config-sip-ua)# transport Configures the SIP user agent (sip-ua) for SIP signaling messages. {udp | tcp} The default value is udp. • udp—Configures the SIP user agent to receive SIP messages on UDP port 5060. • tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060. Cisco IOS Release 12.2(8)T and 12.2(11)T 18
  19. 19. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Command Purpose Step 3 Router(config-sip-ua)# sip-server {dns: Enters the host name or IP address of the SIP server interface. host-name | ipv4:ip-address [port-number]} • dns:—Sets the global SIP server interface to a DNS. • host-name—A valid DNS host name takes the following format: gateway.company.com. • ipv4:ip-address—Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx. • port-number—(Optional) Specifies the port number for the SIP server. Step 4 Router(config-sip-ua)# timers trying Sets time to wait for a response. number • number—Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500. Step 5 Router(config-sip-ua)# timers expires Limits the time duration (in milliseconds) of a search for an number INVITE. • number—Specifies the time (in milliseconds) for which an INVITE request is valid. Possible values are 60000 through 300000. The default is 180000. Step 6 Router(config-sip-ua)# retry invite Configures the SIP signaling timers for retry attempts. number • number—Specifies the number of INVITE retries. Valid values are 1 through 10. The default is 6. Step 7 Router(config-sip-ua)# max-forwards Limits the number of proxy or redirect servers that can forward a number request. • number—Number of hops. Valid values are 1 through 15. The default is 6. Cisco IOS Release 12.2(8)T and 12.2(11)T 19
  20. 20. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Configuring SIP Support for VoIP Dial Peers To configure SIP support for a VoIP dial peer, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# dial-peer voice tag voip Enters dial-peer configuration mode to configure a VoIP dial peer. • tag—Digits that define a particular dial peer. Valid entries are from 1 to 2,147,483,647. Step 2 Router(config-dial-peer)# Defines the telephone number associated with this VoIP dial peer. destination-pattern [+]string[T] • +—(Optional) Character indicating an E.164 standard number. • string—Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: – The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads. On the Cisco 3600 series routers only, these characters cannot be used as leading characters in a string (for example, *650). – Comma (,), which inserts a pause between digits. – Period (.), which matches any entered digit (this character is used as a wildcard). On the Cisco 3600 series routers, the period cannot be used as a leading character in a string (for example, .650). – Percent sign (%), which indicates that the previous digit/pattern occurred zero or multiple times, similar to the wildcard usage in the regular expression. – Plus sign (+), which matches a sequence of one or more matches of the character/pattern. Note The plus sign used as part of the digit string is different from the plus sign that can be used in front of the digit string to indicate that the string is an E.164 standard number. Cisco IOS Release 12.2(8)T and 12.2(11)T 20
  21. 21. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Command Purpose – Circumflex (^), which indicates a match to the beginning of the string. – Dollar sign ($), which matches the null string at the end of the input string. – Backslash symbol (), which is followed by a single character matching that character or used with a single character with no other significance (matching that character). – Question mark (?), which indicates that the previous digit occurred zero or one time. – Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range. This is similar to a regular expression rule. – Parentheses “( )”, which indicate a pattern and is the same as the regular expression rule. • T—(Optional) Control character indicating that the destination-pattern value is a variable length dial string. Step 3 Router(config-dial-peer)# session Enters the session transport type for the SIP user agent. transport {udp | tcp} • udp—Configures the SIP user agent to receive SIP messages on UDP port 5060. • tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060. Step 4 Router(config-dial-peer)# session Enters the session protocol type as IETF Session Inititation protocol sipv2 Protocol. Step 5 Router(config-dial-peer)# session target Specifies the dial peer session target to use the global SIP server. sip-server Configuring a POTS Dial Peer To configure a POTS dial peer, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# dial-peer voice tag voip Enters dial-peer configuration mode to configure a VoIP dial peer. Step 2 Router(config-dial-peer)# Defines the telephone number associated with this POTS dial peer. destination-pattern [+]string[T] Step 3 Router(config-dial-peer)# port Associates this POTS dial peer with a specific voice port. slot-number/subunit-number/port Step 4 Router(config-dial-peer)# session Enters the session transport type for the SIP user agent. transport {udp | tcp} Step 5 Router(config-dial-peer)# session Enters the session protocol type. protocol sipv2 Step 6 Router(config-dial-peer)# session target Specifies the dial peer session target to use the global SIP server. sip-server Cisco IOS Release 12.2(8)T and 12.2(11)T 21
  22. 22. Session Initiation Protocol (SIP) for VoIP Configuration Tasks Configuring SIP Call Transfer for a POTS Dial Peer To configure SIP call transfer for a POTS dial peer, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# dial-peer voice tag pots Enters dial-peer configuration mode to configure a POTS dial peer. Step 2 Router(config-dial-peer)# application Specifies that the standard session application will be invoked for session this dial peer. Step 3 Router(config-dial-peer)# Specifies the telephone number associated with the dial peer. destination-pattern [+]string[T] Step 4 Router(config-dial-peer)# port slot/port Specifies the voice slot number and port through which incoming VoIP calls are received. Configuring SIP Call Transfer for a VoIP Dial Peer To configure SIP call transfer for a VoIP dial peer, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# dial-peer voice tag voip Enters dial-peer configuration mode to configure a VoIP dial peer. Step 2 Router(config-dial-peer)# application Specifies that the standard session application will be invoked for session this dial peer. Step 3 Router(config-dial-peer)# Specifies the telephone number associated with the dial peer. destination-pattern [+]string[T] Step 4 Router(config-dial-peer)# session target Specifies the IP address of the destination gateway for outbound ipv4:ip-address dial peers. Configuring Phone Number Translation Rules By default, the SIP gateway tags called numbers that have 11 or more digits as “international” when sending SETUP messages to the PSTN switch. In some cases, such as situations where the user must dial 9 to access an outside line, this assumption may not be correct. To accommodate such situations, you can define translation rules on the outbound POTS dial peer to convert the “type of number” to the correct value. Translation rules manipulate the called number digits and the “type of number” value associated with the called digits. Cisco IOS Release 12.2(8)T and 12.2(11)T 22
  23. 23. Session Initiation Protocol (SIP) for VoIP Configuration Tasks To define translation rules on a POTS dial peer, enter the following commands beginning in global configuration mode: Command Purpose Step 1 Router(config)# translation-rule name-tag Defines a translation-rule tag number and enters translation-rule configuration mode. All subsequent commands that you enter in this mode before you exit will apply to this translation-rule tag. • name-tag—The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647. Step 2 Router(config-translate)# rule name-tag Specifies translation rules. This command can be entered multiple input-matched-pattern substituted-pattern times and is applied to the translation-rule defined in Step 1. [match-type substituted-type] • name-tag—The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. Range is from 1 through 2,147,483,647. • input-matched-pattern— The input string of digits for which pattern matching is performed. • substituted-pattern—The replacement digit string that results after pattern matching is performed. Regular expressions are used to carry out this process. • match-type—(Optional) The choices for this field are, abbreviated, any, international, national, reserved, subscriber, and unknown, as defined by the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Q.931 specification. If you enter the match-type value, then you must also enter the substituted-type value. • substituted-type—(Optional) The choices for this field are abbreviated, international, national, reserved, subscriber, and unknown, as defined by the ITU Q.931 specification. Step 3 Router(config-translate)# exit Exits from translate configuration mode. Step 4 Router(config)# dial-peer voice tag pots Enter the dial-peer mode to configure a POTS dial peer. Step 5 Router(config-dial-peer)# Specifies the translation tag for an outbound called number. translate-outgoing called name-tag • name-tag—Translation rule tag. Valid values are 1 to 2,147,483,647. Step 6 Router(config-dial-peer)# port Specifies the voice port. slot-number/port For more information about the commands used to configure translation rules, see the Dial Peer Enhancements documentation on Cisco.com. Verifying the SIP Feature Configuration Enter the show running configuration command to verify your configuration. Cisco IOS Release 12.2(8)T and 12.2(11)T 23
  24. 24. Session Initiation Protocol (SIP) for VoIP Configuration Examples Configuration Examples This section provides the following configuration examples: • Basic SIP Configuration Example • Configuring SIP with Multiple Codecs Example • Configuring Phone Number Translation Rules Examples • Call Transfer Configuration Examples Call Transfer Capabilities Using the Refer Method For configuration examples for this feature, see the document Call Transfer Capabilities Using the Refer Method. Configurable PSTN Cause Code to SIP Response Mapping For configuration examples for this feature, see the document Configurable PSTN Cause Code to SIP Response Mapping. Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events For configuration examples for this feature, see the document Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events. ISDN Progress Indicator Support for SIP Using 183 Session Progress There are no configuration examples for this feature. RFC 2782 Compliance for DNS SRV For configuration examples for this feature, see the document SIP Gateway Support of RSVP and TEL URL. SIP Diversion Header Implementation for Redirecting Number For configuration examples for this feature, see the document SIP Diversion Header Implementation for Redirecting Number. SIP Gateway Support for Bind Command For configuration examples for this feature, see the document SIP Gateway Support for Bind Command. SIP Gateway Support of RSVP and TEL URL For configuration examples for this feature, see the document SIP Gateway Support of RSVP and TEL URL. SIP Intra-Gateway Hairpinning There are no configuration examples for this feature. SIP INVITE Request with Malformed Via Header There are no configuration examples for this feature. Cisco IOS Release 12.2(8)T and 12.2(11)T 24
  25. 25. Session Initiation Protocol (SIP) for VoIP Configuration Examples SIP T.38 Fax Relay For configuration examples for this feature, see the document SIP T.38 Fax Relay. SIP User Agent MIB There are no configuration examples for this feature. Basic SIP Configuration Example The following shows an example of the output that appears when you enter the show running configuration command. Router1# show running configuration Building configuration... Current configuration: ! version 12.2 service timestamps debug datetime service timestamps log uptime no service password-encryption ! hostname router1 ! enable secret 5 $1$dlEK$ziROgcQm08RwI/d0VSfal1 enable password password1 ! dspint DSPfarm1/0 ! ip subnet-zero ip tcp path-mtu-discovery ip name-server 172.18.192.48 ! isdn voice-call-failure 0 ! ! controller T1 1/0 framing esf clock source line primary linecode b8zs ! controller T1 1/1 ! ! voice-port 2/0/0 ! voice-port 2/0/1 ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g723r63 codec preference 3 g723r53 !! dial-peer voice 100 pots destination-pattern 3660110 port 2/0/0 ! dial-peer voice 200 pots application session destination-pattern 3660120 Cisco IOS Release 12.2(8)T and 12.2(11)T 25
  26. 26. Session Initiation Protocol (SIP) for VoIP Configuration Examples port 2/0/1 ! dial-peer voice 101 voip destination-pattern 3660210 session protocol sipv2 session target ipv4:166.34.244.73 codec g711ulaw ! dial-peer voice 201 voip application sesion destination-pattern 3660220 session protocol sipv2 session target dns:3660-2.sip.com codec g711ulaw ! dial-peer voice 999 voip destination-pattern 5551111 session protocol sipv2 session target ipv4:161.44.53.89 session transport tcp ! dial-peer voice 300 pots destination-pattern 2101100 ! dial-peer voice 350 voip destination-pattern 3100607 session protocol sipv2 session target ipv4:172.18.192.197 codec g711ulaw ! dial-peer voice 301 voip application session destination-pattern 1234 session protocol sipv2 session target ipv4:172.18.192.193 codec g711ulaw ! dial-peer voice 333 voip application session destination-pattern 1235 session protocol sipv2 session target ipv4:172.18.192.199 codec g711ulaw ! dial-peer voice 888 voip destination-pattern 888 session protocol sipv2 session target ipv4:161.44.53.89 session transport tcp codec g711ulaw ! dial-peer voice 260011 voip destination-pattern 260011 session protocol sipv2 session target ipv4:172.18.192.164 codec g711ulaw ! dial-peer voice 444 voip destination-pattern 2339000 session protocol sipv2 session target ipv4:172.18.192.205 codec g711ulaw ! dial-peer voice 111 voip Cisco IOS Release 12.2(8)T and 12.2(11)T 26
  27. 27. Session Initiation Protocol (SIP) for VoIP Configuration Examples destination-pattern 111 session protocol sipv2 session target sip-server codec g711ulaw ! dial-peer voice 7777777 voip destination-pattern 19197777777 session protocol sipv2 session target ipv4:172.18.192.38 codec g711ulaw ! ! sip-ua max-forwards 0 retry invite 5 retry response 0 retry bye 0 retry cancel 0 retry prack 0 retry comet 0 retry rel1xx 0 retry notify 0 timers trying 501 timers expires 0 timers connect 0 timers disconnect 0 timers prack 0 timers comet 0 timers rel1xx 0 timers notify 0 sip-server ipv4:172.16.0.0 no transport tcp! ! interface FastEthernet0/0 ip address 172.18.192.194 255.255.255.0 load-interval 30 speed auto half-duplex ! interface FastEthernet0/1 ip address 166.34.245.230 255.255.255.224 load-interval 30 speed auto half-duplex ! ip classless ip route 0.0.0.0 0.0.0.0 172.18.192.1 ip route 166.34.0.0 255.255.0.0 166.34.245.225 no ip http server ! access-list 101 permit ip host 10.0.2.30 host 10.0.2.31 access-list 101 deny udp any eq rip any access-list 101 deny udp any any eq rip access-list 101 deny udp any eq isakmp any access-list 101 deny udp any any eq isakmp access-list 101 permit ip any any snmp-server engineID local 000000090200003094202740 snmp-server community public RW ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 Cisco IOS Release 12.2(8)T and 12.2(11)T 27

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