SIP Overview

Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for multimedia conferences 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 a IMG 2020 can be used as a Media Gateway to allow two separate networks to connect. The IMG 2020 supports messaging between SS7 and SIP, and ISDN and SIP. Transcoding from SIP to SIP is also supported. Below is typical diagram of the IMG 2020 in a TDM to IP network.

Supported SIP Features

Configuring SIP

Refer to the Configure SIP (Single SIP IP) and Configure SIP (Multiple SIP IP) topic. 

Basic 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 2020 does not support inbound registration. Inbound Registration is not applicable with Media Gateways. For more information refer to the SIP Signaling - SIP topic.

• Call Release Data in the Radius CDR. Refer to the Call Party Release Source in RADIUS CDR topic for more information.

• Vocoder Data in the Radius CDR. Refer to the RADIUS - Codec Info in CDR topic for more information.

• The IMG 2020 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.

• Create multiple SIP profiles. A SIP Profile allows a user to easily assign a number of SIP features to the IMG 2020. A SGP Profile is first created and then assigned profiles to a gateway in the External Gateway pane. Refer to the SIP Profile - SGP link.

• Supports SIP Response Messages 1xx, 2xx, 3xx, 4xx, 5xx, 6xx

• SIP Session Timer. Refer to the SIP Profile - Session Timer topic for more information.

• SIP Redirection. Refer to the SIP Redirect and SIP Redirect - Initiated 302 Response topics for more information.

• SIP Transcoding. Refer to the Transcoding link for more information.

• SIP Call Hold. Click on SIP Call Hold link for more information.
   

Supported Methods

• SIP INFO Method - RFC 2976.  Refer to SIP INFO Method - DTMF, SIP INFO METHOD - Subscribe/Notify, or SIP INFO Method - Long Call Duration Auditing topics for more information.

• SIP INVITE/ReINVITE Method - RFC 3261 Refer to the SIP INVITE Method topic for more information.

• SIP SUBSCRIBE/NOTIFY Method - RFC 3265. Refer to the SIP INFO METHOD - Subscribe/Notify topic for more information.

• SIP UPDATE Method - RFC 3311. Refer to the SIP Update Method for more information.

• SIP OPTIONS (BUSY OUT) - RFC 3261. Refer to the SIP Options - Busy Out/Keep Alive Configure topic for more information.

• SIP PROXY Registration. Refer to the SIP Profile - Proxy topic for more information.

• SIP CANCEL and BYE Methods - RFC3261

• SIP PRACK (Provisional Response Acknowledgement). Refer to the SIP PRACK reliable provisional responses topic for more information.
   

SIP Extensions

• SIP Diversion Header. Refer to the SIP Diversion Header topic for more information.

• SIP Reason Header. Refer to the SIP Reason Header topic for more information (SIP to TDM and TDM to SIP).

• SIP Privacy Header/Network Identity Header. RFC 3323 and 3325. Refer to the SIP Privacy topic for more information.

• SIP Session Timers. RFC 2543 and 4028. Refer to the SIP Profile - Session Timer topic for more information.

• SIP 3PCC (Third Party Call Control)

• Non-Standard Tags in From/To Header.

• Non-Standard Tags in R-URI.

• SIP History-Info Header Support. 

Routing/Call Handling

• ENUM Support for SIP (RFC 3762).  Refer to the SIP Enum Support topic for more information.

• SIP Load Balancing. Refer to the SIP Load Balancing topic for more information.

• SIP Trunk Group Selection (RFC 4904). Refer to the SIP Profile - Trunk Group Selection topic for more information.

• SIP Proxy Handling. Refer to the SIP Profile - Proxy topic for more information.

• SIP Redirect Server. Refer to the SIP Redirect Server Support topic for more information.

• SIP DNS lookup. The IMG 2020 can route SIP traffic to a remote entity based on the IP Address or the Host Name. Refer to the Configure DNS topic for more information.

• SIP DNS Redundancy. The IMG 2020 supports having multiple DNS servers for redundancy and reliability purposes. Refer to the DNS Server and DNS Client topics for more information.

• Re-origination. This feature limits the number of INVITE re-transmission attempts (1-5 attempts). The number configured supersedes the standard # of re-transmissions specified in RFC 3261 (which is based on timers T1 and T2. The default is Re-transmit All. This feature is enabled in the SIP Profile - SGP.

• SIP Gateway Busy Out. Refer to the SIP Options - Busy Out/Keep Alive Configure for more information.

• DTMF out-of-band transfer using SIP INFO METHOD - Subscribe/Notify.

• Representing trunk groups in SIP Uniform Resource Identifiers (URIs). 

Media

• Network Address Translation (NAT) Traversal. Refer to the Symmetric NAT Traversal topic for more information.

• Relay for Dual Tone Multi Frequency (DTMF) digits, including payload type negotiation (RFC 2833).

• Codec Negotiation Priority. Refer to the SIP Codec Negotiation topic for more information.

  • T.38 Fax Support. Refer to the SIP Fax / Modem Support for more information.

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.

• Supports AnnexB in SDP. Refer to the G.729 AnnexB section of the Vocoder Profile topic for more information.

• G.726 Support for SIP. Refer to the Vocoder Profile Entry. 

Interworking

Basic Support

• SIP-T Support (Session Initiation Protocol for Telephones IETF - RFC 3372). Refer to the Configure SIP to SIP-T Interworking topic for more information.

• 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 SS7 ISUP Interworking (10.5.0 +) RFC 3398.

• 3GPP TS 29.163 Support for SIP to ISUP Interworking

  • SIP Precondition and SS7 Continuity Check Protocol Interworking Support

  • SIP P-Early-Media and SS7 UID Parameters Interworking Support

• ITX (SPIROU) Support SPIROU/ITX in SIP INFO

• Originating Line Info Support

• SIP UUI Support 

SIP

• 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