Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call whereby an IMG can be used as a Media Gateway to allow two separate networks to connect. The IMG supports SS7 to SIP, ISDN to SIP, CAS to SIP, SIP to SIP, and H.323 to SIP. Below is exemplary diagram of an IMG in a TDM to IP network.
RFC |
Description |
2246 |
Transport Layer Security (TLS) for SIP |
2327 |
Session Description Protocol (SDP) |
2976 |
SIP Info |
3240 |
Internet media type message/sipfrag |
3261 |
SIP: Session Initiation Protocol |
3262 |
SIP PRACK |
3263 |
Locating SIP servers for DNS lookup SRV and A records |
3264 |
SDP Offer/Answer Model (Do not support multiple 'm' lines in SIP SDP) |
3265 |
SIP Subscribe/Notify |
3311 |
SIP Update |
3323 |
SIP Privacy Header |
3325 |
Asserted Identity |
3326 |
SIP Reason Header |
3332 |
M3UA Adaption Layer |
3372 |
SIP for Telephones (SIP-T/SIP-I) |
3398 |
ISUP/SIP Mapping |
3515 |
SIP Refer |
3551 |
Payload Type Support |
3578 |
ISUP Overlap Signaling to SIP |
3581 |
Symmetric Response Routing |
3666 |
Call Flows - SIP to PSTN Dialing |
3711 |
IP Media Layer Security Standard (RTP/RTCP) |
3725 |
Third Party Call Control for SIP |
3764 |
ENUM for SIP Address of Record |
3891 |
SIP Replace Header |
3892 |
SIP Referred by Mechanism |
4028 |
SIP Session Timer |
4040 |
Clear Channel Codec Support |
4244 |
SIP History info (for call diversion) |
4568 |
IP Signaling Layer Security Standard (RTP/RTCP) |
4904 |
Trunk Group Parameter Support |
RFC 3261, SIP (Session Initiation Protocol)
Backward compatible with entities running RFC 2543
RFC 3581, Rport Extension Parameter in the Via Header
RFC 2327 SDP Support
RFC 3551 RTP Profile for Audio and Video Conferees with minimal control.
RFC 3666 Call Flows -- SIP to PSTN Dialing
RFC 3960 - Early Media and Ringing Tone Generation
Transmission Control Protocol Support (TCP/IP). Single or multi-socket use.
Reliable User Datagram Protocol (UDP) transport, with retransmissions
SIP Authentication and Outbound Registration. The IMG does not support inbound registration. Inbound Registration is not applicable with Media Gateways. For more information click on Authentication Link
Call Release Data in the Radius CDR (Software 10.5.0 +) Click HERE for more information
Vocoder Data in the Radius CDR (Software 10.5.0 +). Click HERE for more information
The IMG supports being a User Agent Client (UAC) or User Agent Server (UAS) and will inter-operate with SIP proxies.
Supports Early Media. Supports 180/183 Session Progress. Click HERE for more information.
Can create multiple SIP profiles. A SIP Profile allows you to easily assign a number of SIP features to a Physical IMG. You create a SIP Profile and then assign profiles to a gateway in the External Gateway pane. See SIP Profiles link.
Supported Response Messages 1xx, 2xx, 3xx, 4xx, 5xx, 6xx
SIP Session Timer. Click on Session Timer link for more information.
SIP Redirection. Click on SIP Redirect Server Support link for more information
SIP Transcoding. Click on Transcoding via SIP link for more information.
SIP Call Hold. Click on SIP Call Hold link for more information.
SIP INFO Method - RFC 2976. See SIP INFO for more information.
SIP INVITE/ReINVITE Method - RFC 3261 See SIP INVITE for more information.
SIP SUBSCRIBE/NOTIFY Method. - RFC 3265. See SUBSCRIBE/NOTIFY for more information.
SIP UPDATE Method - RFC 3311. See SIP UPDATE for more information
SIP OPTIONS (BUSY OUT) - RFC 3261. See SIP Options Keep Alive link for more information
SIP PROXY Registration See SIP Proxy Handling and SIP Proxy link for more information REGISTER (Outbound)
SIP CANCEL and BYE Methods - RFC3261
SIP PRACK (Provisional Response Acknowledgement) See SIP PRACK reliable provisional responses
SIP Diversion Header. See SIP Diversion Header link for more information
SIP Reason Header. See SIP Reason Header link for more information. (SIP to TDM and TDM to SIP)
SIP Privacy Header/Network Identity Header. RFC 3323 AND 3325. See SIP Privacy Headers, Configuring SIP Privacy, and Remote party ID for more information.
SIP Session Timers. RFC 2543 and 4028. See SIP Session Timer for more information.
SIP 3PCC. See SIP 3rd Party Call Control (3PCC) for more information.
Non-Standard Tags in From/To Header
Non-Standard Tags in R-URI
SIP History-Info Header Support
ENUM Support for SIP. RFC 3762 See ENUM link for more information
SIP Load Balancing. See SIP-Based Load Balancing/Virtual IP Address for more information
SIP Trunk Group Selection. See SIP Trunk Group Selection for more information
SIP Proxy Handling. See SIP Proxy Handling for more information.
SIP Redirect Server. See SIP Redirect Server Support and 3xx responses
SIP DNS Lookup. The IMG can route SIP traffic to a remote entity based on the IP Address or the Host Name. DNS (Domain Name Server) Lookup
SIP DNS Redundancy. The IMG supports having multiple DNS servers for redundancy and reliability purposes. See DNS Server and DNS Client panes.
Re-origination. This feature allows you to limit the number of INVITE re-transmission attempts (1-5 attempts). The number configured supersedes the standard # of re-transmissions specified in RFC3261 (which is based on timers T1 and T2. The default is Re-transmit All. You enable this feature in the SIP Profile.
SIP Gateway Busy Out. See SIP Gateway Busy Out for more information
DTMF out-of-band transfer using INFO and SUBSCRIBE/NOTIFY
Representing trunk groups in SIP Uniform Resource Identifiers (URIs)
SIP PRACK in 1xx messages. See SIP PRACK reliable provisional responses
SIP Trunk Group Selection RFC 4904
Network Address Translation (NAT) Traversal. See Symmetric NAT Traversal for more information
Relay for Dual Tone Multi Frequency (DTMF) digits, including payload type negotiation (RFC 2833)
Codec Negotiation Priority. See SIP Codec Negotiation Priority Selection for more information
T.38 Fax support. See T.38 Real Time Fax using SIP for more information.
Support for RFC 3550 (RTP: A Transport Protocol for Real-Time Applications) – Partially compliant.
Note: If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message, the gateway returns the same rate in the response SDP.
Fax Modem Support. See Modem Support for more information.
Receive and Transmit Gain control on per channel basis. See Gain Control on SIP Channel Groups
Supports AnnexB in SDP. See G.729 AnnexB Selection for more information
GSM Support for SIP (10.5.0 +). See Setting Host Flags , Supported Codecs , Vocoder Entry , and Vocoder Information links for more information.
G.726 Support for SIP (10.5.0 +). See Setting Host Flags , Supported Codecs , Vocoder Entry , and Vocoder Information for more information
Basic Support:
SIP-T Support (Session Initiation Protocol for Telephones IETF) RFC 3372 - See and Configuring SIP-T links.
SIP-I Support (Session Initiation Protocol for Telephones ITU ) RFC 3372
RFC 3666 Call Flows -- SIP to PSTN Dialing
Q.1912.5 Support for SIP to ISUP Interworking (10.5.0 +) RFC 3398.
SS7 to SIP Calling Party Category (CPC)
SIP UUI support
SIP to CAS
SIP 3xx Gateway Responses
SIP Diversion Header
SIP Trunk Group ID's
SIP Codec Negotiation
SIP Busy Out Modem BypasS
SIP-T
SIP-I
SIP over TLS
SIP Refer
SIP Refer for Call Transfer
SIP Dial Around Indicator Support
Generic Name Indicator (SIP to SS7)
M3UA Signaling Gateway for TCAP/SCCP