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Table of Contents

Cisco SIP Compliance Reference Information
 SIP Functions

SIP Methods
 SIP Responses
 SIP Header Fields

SIP Transport Layer Protocols

SIP Security
 SIP Session Description Protocol (SDP) Usage

SIP DNS Records Usage

SIP DTMF Relay
 Session Timer Draft
 Fax Relay
 SIP T.37/T.38 Fax Relay
 SIP synchronization with RSVP for QoS

Cisco SIP Compliance Reference Information

This section describes how the Cisco SIP User Agent and the Cisco SIP Gateway comply with the IETF definition of SIP as described in RFC 2543.

This section contains compliance information on the following:

SIP Functions

Function  Supported?  Comments 

User Agent Client (UAC)

Yes

 

User Agent Server (UAS)

Yes

 

Proxy Server/Redirect Server/Registrar

 

No.

The SIP gateway does not have the proxy or redirect server functionality, but can work with an external proxy or redirect server.

SIP Methods

There are six methods used by the SIP gateway.

Method  Supported?  Comments 

INVITE

Yes

The gateway supports mid-call INVITEs with the same call ID but different SDP session parameters (to change the transport address). Mid-call Invite can change other parameters of the SDP, such as port number or codec. The gateway also supports mid-call INVITEs for refreshing session timer value.

ACK

Yes

 

OPTIONS

Yes

The gateway does not generate OPTIONS. However, it will respond to OPTIONS requests.

BYE

Yes

 

CANCEL

Yes

 

COMET

Yes

Conditions MET. Used in QoS implementation to indicate to other endpoint whether or not the conditions have been met, i.e. resources reserved.

NOTIFY

Yes

Used in implementation of REFER to let initiator of the REFER know the outcome of the transfer. It is also used to notify a subscriber of any change in events to which a subscriber is subscribed to, such as dtmf-events, mwi, etc.

PRACK

Yes

Provisional ACKnowledgement. Used in reliable provisional responses.

REFER

Yes

The gateway responds to a REFER. It generates REFER for call transfer (attended and blind)..

REGISTER

No

 

SUBSCRIBE

Yes

Gateway can generate and accept SUBSCRIBE. It processes SUBSCRIBE for selected applications such as telephony events (DTMF) and for Message Waiting Indication (MWI).

INFO

Yes

Gateway can generate and accept INFO.

UPDATE

Yes

Gateway can accept and modify an early or established dialog.

SIP Responses

Cisco IOS Release 12.2(10)T supports the following SIP Responses:

1xx Response—Information Responses

1xx Response  Comments 

100 Trying

This response indicates that action is being taken on behalf of the caller, but that the callee has not yet been located.

The SIP gateway generates this response for an incoming INVITE.

Upon receiving this response, the gateway stops retransmitting INVITEs. It then waits for a 180 Ringing or 200 OK response.

180 Ringing

This response indicates that the callee has been located and is being notified of the call.

The SIP gateway generates a 180 Ringing response when the called party has been located and is being alerted.

Upon receiving this response, the gateway generates local ringback, then it waits for a 200 OK response.

181 Call is being forwarded

This response indicates that the call is being rerouted to another destination.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway processes the responses the same way that it processes a 100 Trying response.

182 Queued

This response indicates that the callee is not currently available but that they have elected to queue the call rather than reject it.

 

183 Session progress

This response is used to perform inband alerting for the caller.

The SIP gateway generates a 183 Session progress response when it receives an ISDN Progress message with an appropriate media indication from PSTN.

2xx Response—Successful Responses

2xx Response  Comments 

200 OK

This response indicates that the request has been successfully processed. The action taken depends on the request made.

The SIP gateway generates this response when the PBX indicates that the user has answered the phone.

Upon receiving this response, gateway sends Connect message to PSTN/PBX and responds with ACK on the IP leg.

3xx Response—Redirection Responses

3xx Response  Comments 

300 Multiple choices

This response indicates that the address resolved to more than one location. All locations are provided and the user or UA is allowed to select which location to use.

The SIP gateway does not generate this response. Upon receiving this response, the gateway contacts the new address in the Contact header field.

301 Moved permanently

This response indicates that the user is no longer available at the specified location. An alternate location is included in the header.

302 Moved temporarily

This response indicates that the user is temporarily unavailable at the specified location. An alternate location is included in the header.

305 Use proxy

This response indicates that the caller must use a proxy to contact the callee.

380 Alternative service

This response indicates that the call was unsuccessful, but that alternative services are available.

4xx Response—Request Failure Responses

4xx Response  Comments 

400 Bad Request

This response indicates that the request could not be understood because of an illegal format.

The SIP gateway generates a 400 Bad Request response for a badly formed request.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

401 Unauthorized

This response indicates that the request requires user authentication.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

402 Payment required

This response indicates that payment is required to complete the call.

403 Forbidden

This response indicates that the server has received and understood the request but will not provide the service.

404 Not Found

This response indicates that the server has definite information that the user does not exist in the specified domain.

The SIP gateway generates this response if it is unable to locate the callee.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

405 Method Not Allowed

This response indicates that the method specified in the request is not allowed. The response contains a list of allowed methods.

The SIP gateway generates this response if an invalid method is specified in the request.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

406 Not Acceptable

This response indicates that the requested resource is capable of generating only responses that have content characteristics not acceptable as specified in the accept header of the request.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

407 Proxy authentication required

This response is similar to the 401 Unauthorized response. However, this response indicates that the client must first authenticate itself with the proxy.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

408 Request timeout

This response indicates that the server could not produce a response before the Expires time out.

409 Conflict

This response indicates that the request could not be processed because of a conflict with the current state of the resource.

410 Gone

This response indicates that a resource is no longer available at the server and no forwarding address is known.

The SIP gateway generates this response if the PSTN returns a cause code of unallocated number.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

411 Length required

This response indicates that the user refuses to accept the request without a defined content length.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

413 Request entity too large

This response indicates that server refuses to process the request because it is larger than the server is willing or able to process.

414 Request-URI too long

This response indicates that the server refuses to process the request because the Request-URI is too long for the server to interpret.

415 Unsupported media

This response indicates that the server refuses to process the request because the format of the body is not supported by the destination endpoint.

The SIP gateway generates this response when it gets an Info for an unsupported event-type. Supported event types are 0-9, A-D, # and *

420 Bad extension

This response indicates that the server could not understand the protocol extension indicated in the Require header.

The SIP gateway generates this response if it cannot understand the service requested in the Require header field.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

480 Temporarily unavailable

This response indicates that the callee was contacted but is temporarily unavailable.

The SIP gateway generates this response if the callee is unavailable (for example, the callee does not answer the phone within a certain amount of time or the called number does not exist or is no longer in service).

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

481 Call leg/transaction does not exist

This response indicates that the server is ignoring the request because it was either a BYE for which there was no matching leg ID or a CANCEL for which there was no matching transaction.

The SIP gateway generates this response if the call leg ID or transaction cannot be identified.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

482 Loop detected

This response indicates that the server received a request that included itself in the path.

Gateway generates this response when it detects the same request has arrived more than once in different paths (most likely due to forking).

483 Too many hops

This response indicates that the server received a request that required more hops than allowed by the Max-Forwards header.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

 

484 Address incomplete

This response indicates that the server received a request containing an incomplete address.

485 Ambiguous

This response indicates that the server received a request in which the callee address was ambiguous. It can provide possible alternate addresses.

486 Busy here

This response indicates that the callee was contacted but that their system is unable to take additional calls.

The SIP gateway generates this response if the callee was contacted but was busy.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

487 Request cancelled

This response indicates that the request was terminated by a BYE or CANCEL request.

The SIP gateway generates this response to an unexpected BYE or CANCEL received for a request.

488 Not Acceptable Media

This response indicates an error in handling the request at this time.

The SIP gateway generates this response if the media negotiation fails.

422 Session Timer Too Small

It is generated by the gateway (UAS) when a request contains a Session-Expires header with a duration that is below the minimum timer for the gateway server. The 422 response MUST contain a Min-SE header with a minimum timer for that server.

491 Request Pending

Gateway generates this error response to reject an UPDATE with new offer, before it receives answer sdp to an offer it has generated in a previous request (Invite or Update).

5xx Response—Server Failure Responses

5xx Response  Comments 

500 Server internal error

This response indicates that the server or gateway encountered an unexpected error that prevented it from processing the request.

The SIP gateway generates this response if it encountered an unexpected error that prevented it from processing the request.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

501 Not implemented

This response indicates that the server or gateway does not support the functions required to complete the request.

502 Bad gateway

This response indicates that the server or gateway received an invalid response from a downstream server.

503 Service unavailable

This response indicates that the server or gateway is unable to process the request due to an overload or maintenance problem.

504 Gateway timeout

This response indicates that the server or gateway did not receive a timely response from another server (such as a location server).

505 Version not supported

This response indicates that the server or gateway does not support the version of the SIP protocol used in the request.

580 Precondition failed

The SIP gateway uses this response code to indicate a failure in having Qos preconditions met for a call.

6xx Response—Global Responses

6xx Response  Comments 

600 Busy everywhere

This response indicates that the callee was contacted but that the callee is busy and cannot take the call at this time.

The SIP gateway does not generate this response.

Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.

603 Decline

This response indicates that the callee was contacted but cannot or does not want to participate in the call.

604 Does not exist anywhere

This response indicates that the server has authoritative information that the callee does not exist in the network.

606 Not acceptable

This response indicates that the callee was contacted, but that some aspect of the session description was unacceptable.

 

SIP Header Fields

Header Field  Supported? 

Accept

Yes

Accept-Encoding

No

Accept-Language

Yes

Allow

Yes

Also

Yes

Authorization

No

Call-ID

Yes

Contact

Yes

Content-Encoding

No

Content-Length

Yes

Content-Disposition

Yes

Content-Type

Yes

Cseq

Yes

Date

Yes

Diversion

Yes

Encryption

No

Event

Yes

Expires

Yes

From

Yes

Hide

No

Max-Forwards

Yes

Organization

No

Min-SE

Yes

Priority

No

Proxy-Authenticate

No

Proxy Authorization

Yes

Proxy-Require

No

RAcr

Yes

RSeq

Yes

Record-Route

Yes

Require

Yes

Response-Key

No

Retry-After

No

Refer-to

Yes

Referred-By

Yes

Replaces

Yes

Requested-By

Yes

Route

Yes

Server

Yes

Session-Expires

Yes

Subject

No

Supported

Yes

Timestamp

Yes

To

Yes

Unsupported

Yes

User-Agent

Yes

Via

Yes

Warning

Yes

WWW-Authenticate

No

SIP Transport Layer Protocols

Transport Layer Protocol  Supported?   Comments  

Unicast UDP

Yes

None.

Multicast UDP

No

 

TCP

Yes

None.

SIP Security

Encryption

Encryption Mode  Supported?  Comments 

End-to-end Encryption

No

IPSEC can be used for security.

Privacy of SIP Responses

No

None.

Hop-by-Hope Encryption

No

IPSEC can be used for security.

Via Field Encryption

No

Authentication

Encryption Mode  No 

Basic Authentication

No

Digest Authentication

No

Proxy Authentication

No

PGP

No

SIP Session Description Protocol (SDP) Usage

SDP Headers  Supported? 

v—Protocol version

Yes

o—Owner/creator and session identifier

Yes

s—Session name

Yes

c—Connection information

Yes

t—Time the session is active

Yes

m—Media name and transport address

Yes

a—Attribute. The primary means for extending SDP and tailoring it to particular applications or media.

Yes

SIP DNS Records Usage

DNS Resource Record Type  Supported? 

Type A

Yes

Type SRV

Yes. Both RFC2052 and RFC2782 formatting are supported.

SIP DTMF Relay

Cisco gateways support DTMF Relay in accordance with RFC2833. The DTMF Relay method is defined as "rtp-nte" in gateway configuration. It is based on the transmission of Named Telephony Events (NTE) and DTMF digits over RTP stream. The default RTP payload type for rtp-nte is 101. The default method of dtmf-relay is inband voice

Cisco gateways also support cisco-rtp which is a cisco proprietary method.

Methods  Supported? 

RFC 2833 AVT Tones

Yes

Cisco RTP Method

Yes

Session Timer Draft

This supports draft-ietf-sip-session-timer-08.txt

Fields  Supported? 

Supported

Yes

Note   Cisco gateways do not initiate use of keepalives for any calls that it originates or terminates. If the "far side" wants to use keepalives, the gateway complies.

Fax Relay

Methods  Supported?  Comment 

T.38 fax relay

Yes

 

Cisco Fax relay

Yes

Not supported on 5350 and 5400 platform

SIP T.37/T.38 Fax Relay

Cisco gateways support T.38/T.37 fax relay/store and forward mechanisms. The following table lists requirements based on Annex-D of ITU recommendation T.38. It indicates whether the requirements are supported.

Requirement  Description  Mandatory/Optional  Supported 

<SIPt38-01>

T.38 over SIP shall be implemented as described in ANNEX D to Recommendation T.38 - SIP/SDP Call Establishment Procedures (previously known as TD0262rev2)

Mandatory

Yes

<SIPt38-02>

SIP-enabled VoIP gateways must be able to detect the CNG, CED fax tones and/or the preamble flag sequence transmitted inside the audio RTP streams

Mandatory

Yes (CED, V.21 preamble detected now. Future DSP releases will detect CNG. CNG is an optional tone.)

<SIPt38-03>

Detection of a fax transmission MUST be performed by the receiving gateway by recognizing the CED tone

Mandatory

Yes

<SIPt38-04>

If CED is not present, fax transmission MUST be detected by the receiving gateway by recognizing the Preamble flag sequence

Mandatory

Yes

<SIPt38-05>

Upon detection of the fax transmission, the gateway MUST initiate switch over to T.38 fax mode by sending a re-INVITE with proper SDP description

Mandatory

Yes

<SIPt38-06>

To prevent glare, even if the transmitting gateway detects the fax transmission (CNG tone), it MUST not initiate the switch over to T.38 fax mode

Mandatory

Yes

<SIPt38-07>

If a SIP session starts with audio capabilities, then switches to fax, the capability must be provided for the session to switch back to audio mode at the end of the fax transmission

Mandatory

Yes

<SIPt38-08>

Support of SIP T.38 fax calls over TCP is desired

Desirable

UDP only

<SIPt38-09>

The SDP protocol type UDPTL (Facsimile UDP transport Layer) must be supported as lately defined by IANA

Mandatory

Yes

<SIPt38-10>

Following SDP attributes as defined by IANA to support T.38 fax sessions must be incorporated:

Registered SDP Protocol format , MIME media type image/t38:

  • MIME media type name: image
  • MIME subtype name: t38

Mandatory

Yes

<SIPt38-11>

The following SDP attributes as defined by IANA to support T.38 sessions must be incorporated.

1. T38FaxVersion

2. T38maxBitRate

3. T38FaxFillBitRemoval

4. T38FaxTranscodingMMR

5. T38FaxTranscodingJBIG

6. T38FaxRateManagement

7. T38FaxMaxBuffer

8. T38FaxMaxDatagram

9. T38FaxUdpEC

Mandatory

Yes

<SIPt38-12>

Any SIP-enabled Cisco gateway supporting T.38 must be able to interoperate with gateways from Cisco and other vendors.

Mandatory

Yes

<SIPt38-13>

Interoperability with gateways supporting T.38 over H.323 is desired and should be addressed as part of overall SIP H.323 interoperability implementation.

Optional

No

<SIPt38-14>

Configuration of SIP enabled gateways should include management of SIP T.38 specific configurable choices.

Mandatory

Yes. The following are configurable:

  • bitrate
  • TCP/UDP (UDP only)
  • hs, ls redundancy
  • ECM

<SIPt38-15>

Tracking and reporting of SIP T.38 activity on the gateways is desired. This includes generation of Call Detail Records (CDR) for SIP T.38 fax calls.

Mandatory

Yes

<SIPt38-16>

The security mechanisms provided in RFC2543 apply. Message authentication can be performed on SIP INVITEs and BYE.

Optional

No

SIP synchronization with RSVP for QoS

This is a multi-source draft dated 7/2000.

Fields  Supported? 

Supported

Yes


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Posted: Wed Oct 1 00:21:47 PDT 2003
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