Session Initiation Protocol, more commonly known as SIP, is a signaling protocol for packet-based networks and is commonly used, along with H.323 to provide signaling for voice over IP (VoIP) communications.
SN10K and SIP
SmartMedia provides support for signaling using the Session Initiation Protocol, more commonly known as SIP, for voice over IP (VoIP) communications. SIP may be used in conjunction with various voice codecs for the media component of a call. Patton SmartMedia media gateways support SIP signaling concurrently with SS7, ISDN and other signaling protocols.
SIP signaling stacks are configured for IP applications and for each SmartMedia unit requiring SIP signaling.
Based upon your system requirements, you can configure a SIP stack to carry signaling traffic over multiple transport servers, which are IP endpoints comprised of: protocol type (TCP/UDP), port number, IP interface, IP address, IP name, and SAPs.
A conceptual illustration is provided below:
Note: The SIP SAPs are hidden by default in the Web Portal configuration. One SIP SAP is automatically allocated for each allocated transport server. An "advanced" mode is available for users which would like to manually assign transport servers to SAPs for some more advanced configurations.
While SmartMedia gateways can perform multiple simultaneous functions such as switching and transcoding as well as deliver value-added services such as IVR or conferencing, they can also be configured to perform a single function. In this case, it is possible to configure SmartMedia gateway to act as a SIP gateway.
Maximum Capacity
Model |
SmartMedia
Release
| SIP SAP | SIP Transport Server | SIP NAP |
|---|
| SN10100 |
2.2-2.5 |
4 |
10 |
256 |
| 2.6+ |
16 |
16 |
512 |
| SN10200 |
2.2-2.5 |
4 |
10 |
256 |
| 2.6+ |
16 |
16 |
512 |
| SN10300 |
2.2-2.5 |
64
(4/SN10300 Unit) |
160
(10/SN10300 Unit) |
256 |
| 2.6+ |
256
(16/SN10300 Unit) |
256
(16/SN10300 Unit) |
512 |
SmartMedia SIP Implementation
SmartMedia SIP implementation works on top of a couple of layers, including SIP and TUCL. In the following figure, grey boxes represent entities that need allocation on the Unit. The TUCL layer is a transport layer used by SIP on our architecture. TUCL presents some advantages over a simple TCP/IP stack. For instance, it adds tracing facilities to any virtual interface.
Supported SIP RFCs
SmartMedia supports the following RFCs for SIP:
| Specification | SmartMedia SIP stack support | API Support | Media Gateway Application Support |
|---|
| RFC 2327 Session Description Protocol |
Yes |
Complete |
Complete |
| RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals |
Yes |
Complete |
Complete |
| RFC 2976 SIP INFO Method |
Yes |
Complete |
Partial (For DTMF Tones Only) |
| RFC 3204 MIME media types for ISUP and QSIG Objects |
Yes |
Complete |
Complete |
| RFC 3261 Session Initiate Protocol |
Yes |
Complete |
Complete |
| RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP) |
Yes |
Complete |
Complete |
| RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers |
Yes |
Complete |
Complete |
| RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP) |
Yes |
Complete |
Partial ('Indicating capabilities' not supported) |
| RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification |
Yes |
No |
No |
| RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method |
Yes |
Partial (Only For Session Timer Refresh) |
Partial (Only For Session Timer Refresh) |
| RFC 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP) |
Yes |
Partial1 |
Partial1 |
| RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks |
Yes |
Partial (No P-Preferred-Identity) |
Partial (No P-Preferred-Identity) |
| RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP) |
Yes |
Complete |
Complete |
| RFC 3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) |
Yes |
Complete |
Complete |
| RFC 3398 ISUP-SIP Mapping |
Yes |
Complete |
Complete |
| RFC 3515 Refer Method |
Yes |
Complete |
Complete |
| RFC 3578 Overlap |
Yes |
Partial (Conversion of ISUP Overlap Signalling into SIP en-bloc Signalling) |
Partial (Conversion of ISUP Overlap Signalling into SIP en-bloc Signalling) |
| RFC 3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing |
Yes |
Complete |
Complete |
| RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples |
Yes |
Partial* |
Partial* |
| RFC 3666 Public Switched Telephone Network (PSTN) Call Flows |
Yes |
Complete |
Complete |
| RFC 3764 enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record |
Yes |
Complete |
Complete |
| RFC 3891 "Replaces" Header |
Yes |
Complete |
Complete |
| RFC 3892 Referred-By Mechanism |
Yes |
No |
No |
| RFC 4028 Session Timers in the Session Initiation Protocol (SIP) |
Yes |
Complete |
Complete |
| RFC 4694 Number Portability Parameters for the "tel" URI |
Yes |
Yes |
Partial (relay of rn and npdi SIP<->SS7) |
| RFC 5806 Diversion Indication in SIP |
Yes |
Yes |
Unconditional forward scenario
|
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