cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/admin
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Table of Contents

SIP Compliance with RFC-2543 Information
SIP Functions

SIP Methods
SIP Responses
SIP Header Fields

SIP Session Description Protocol (SDP) Usage

SIP Compliance with RFC-2543 Information


This section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 2543.

This section contains compliance information on the following:

SIP Functions

Function  Supported? 

User Agent Client (UAC)

Yes

User Agent Server (UAS)

Yes

Proxy Server

Third-party only

Redirect Server

Third-party only

SIP Methods

Five of the six methods used by the SIP gateway are supported:

Method  Supported?  Comments 

INVITE

Yes

The Cisco SIP IP phone supports mid-call changes such as putting a call on hold as signaled by a new INVITE that contains an existing Call-ID.

ACK

Yes

None.

OPTIONS

No

BYE

Yes

CANCEL

Yes

REGISTER

Yes

The Cisco SIP IP phone supports both user and device registration.

SIP Responses

Release 1.0 of the Cisco SIP IP phone supports the following SIP responses:

1xx Response—Information Responses

1xx Response  Supported?  Comments 

100 Trying

Yes

None

180 Ringing

Yes

181 Call is being forwarded

See comments

The Cisco SIP IP phone does not generate these responses, however, the phone does receive them. The phone processes these responses the same way that it processes the 100 Trying response.

182 Queued

183 Session Progress

2xx Response—Successful Responses

2xx Response  Supported?  Comments 

200 OK

Yes

None

3xx Response—Redirection Responses

3xx Response  Supported  Comments 

300 Multiple Choices

Yes

None

301 Moved Permanently

Yes

302 Moved Temporarily

Yes

305 Use Proxy

Yes

The phone does not generate these responses. The gateway contacts the new address in the Contact header field.

380 Alternate Service

Yes

4xx Response—Request Failure Responses

4xx Response  Supported?  Comments 

400 Bad Request

Yes

The phone generates a 400 Bad Request response for a erroneous request. For an incoming response, the phone initiates a graceful call disconnect (during which the caller hears a busy or fast busy tone) before clearing the call request.

401 Unauthorized

Yes

This response is only received in this release.

If a 401 Unauthorized response is received during registration, the phone accepts the response and sends a new request that contains the user's authentication information in the format of the HTTP digest as modified by RFC 2543.

402 Payment Required

Yes

The phone does not generate the 402 Payment Required response.

403 Forbidden

Yes

This response is only received in this release.

If the phone receives a 403 Forbidden response, it notifies the user of the response. This response indicates that the SIP server has the request but will not provide service.

404 Not Found

See comments

This response is only received in this release.

If the phone receives a 403 Forbidden response, it notifies the user of the response. This response indicates that the server has definite information that the user does not exist in the domain specified in the Request-URI header field.

405 Method Not Allowed

See comments

This response is only received in this release.

If the phone receives a 405 Method Not Allowed response, it notifies the user of the response.

406 Not Acceptable

See comments

The SIP phone does not generate a 406 Not Acceptable response. For an incoming response, the gateway initiates a graceful call disconnect (during which the caller hears a busy or fast busy tone) before clearing the call request.

407 Proxy Authentication Required

See comments

This response is only received in this release.

The 407 Proxy Authentication Required response indicates that the phone must first authenticate itself with the proxy server. If received by the phone, the phone may repeat the INVITE request with a suitable Proxy-Authorization field. This field should contain the authentication information of the user agent for the next outbound proxy or gateway.

408 Request Timeout

See comments

The SIP phone does not generate a 408 Request Timeout response. For an incoming response, the gateway initiates a graceful call disconnect (during which the caller hears a busy or fast busy tone) before clearing the call request.

409 Conflict

See comments

This response is only received the phone in this release.

The 409 Conflict response indicates that the INVITE request could not be process because of a conflict with the current state of the resource. If this response is received, the user is notified.

410 Gone

See comments

This response is only received by the phone in this release.

The 410 Gone response indicates that a resource is no longer available at the server and no forwarding address is known.

411 Length Required

See comments

This response is only received by the phone in this release.

This response indicates that the user refuses to accept the request without a defined content length. If received the phone resends the INVITE request if it can add a valid Content-Length header field.

413 Request Entity Too Large

See comments

This response is only received by the phone in this release.

If a retry after header field is contained in this response, then the user can attempt the call once again in the retry time provided.

414 Request—URL Too Long

See comments

This response is only received by the phone in this release. The user is notified if this response is received.

415 Unsupported Media

See comments

This response is only received by the phone in this release. The user is notified if this response is received.

420 Bad Extension

See comments

This response is only received by the phone in this release. The user is notified if this response is received.

If the phone does not understand the protocol extension specified in the Require field, the 420 Bad Extension response is generated.

480 Temporarily Unavailable

See comments

This response is only received by the phone in this release. The user is notified if this response is received.

If this response is received, the user is notified that the callee is temporarily unavailable (perhaps not logged on) and any retry information is displayed.

481 Call Leg/Transaction Does Not Exist

See comments

This response is only received by the phone in this release. The user is notified if this response is received.

482 Loop Detected

483 Too Many Hops

484 Address Incomplete

485 Ambiguous

See comments

This response is only received by the phone in this release.

If a new contact is received, the phone might re-initiate the call.

486 Busy Here

See comments

This response is only received by the phone in this release.

When this response is received, the user is notified and a busy tone is generated.

5xx Response—Server Failure Responses

5xx Response  Comments 

500 Internal Server Error

For an incoming response, the Cisco SIP IP phone sends a new request if an additional contact address is present. If an additional contact address is not present, the gateway initiates a graceful call disconnect.

501 Not Implemented

502 Bad Gateway

503 Service Unavailable

504 Gateway Timeout

505 Version Not Supported

6xx Response—Global Responses

6xx Response  Comments 

600 Busy Everywhere

The Cisco SIP IP phone does not generate these 6xx responses. For an incoming response, the gateway initiates a graceful call disconnect.

603 Decline

604 Does Not Exist Anywhere

606 Not Acceptable

SIP Header Fields

Header Field  Supported? 

Accept

Yes

Accept-Encoding

Yes

Accept-Language

Yes

Allow

Yes

Also

Yes

Authorization

Yes

Call-ID

Yes

Contact

Yes

Content-Encoding

Yes

Content-Length

Yes

Content-Type

Yes

Cseq

Yes

Date

Yes

Encryption

No

Expires

Yes

From

Yes

Hide

No

Max-Forwards

Yes

Organization

No

Priority

No

Proxy-Authenticate

Yes

Proxy-Authorization

Yes

Proxy-Require

Yes

ReBy

Yes

Record-Route

Yes

Require

Yes

Response-Key

No

Retry-After

Yes

Route

Yes

Server

No

Subject

No

Timestamp

Yes

To

Yes

Unsupported

Yes

User-Agent

Yes

Via

Yes

Warning

Yes

WWW-Authenticate

Yes

SIP Session Description Protocol (SDP) Usage

SDP Headers  Supported? 

v—Protocol version

Yes

o—Owner/creator and session identifier

Yes

a—Session name

Yes

c—Connection information

Yes

m—Media name and transport address

Yes


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Posted: Sat Nov 29 14:48:20 PST 2003
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