Summary:ASTERISK-18703: NO acceptance of SDP packets with set i= field
Reporter:xyz312 (xyz312)Labels:
Date Opened:2011-10-10 17:04:08Date Closed:2015-03-14 10:19:12
Versions:1.8.4 Frequency of
Environment:Linux x61 2.6.38-8-generic #42-Ubuntu 1: Attachments:
Description:I tried a few days to send a message with the help of the i= field in the SDP part of an SIP packet during a call setup from a Caller to a Callee.
According to RFC 4566 the i= field can be used for:

"The "i=" field is intended to provide a free-form human-readable description of the session or the purpose of a media stream.  It is not suitable for parsing by automata."

However the Asterisk server does not accept the INVITE packet when the i= field is set. The same call without the i= field does work properly. Furthermore in the debug/verbose output (set verbose/debug 4 and sip debug to on according to the bug tracking rules in concern to SIP) does not show me any information about the acceptance of the packet with the i= field. It is just ignored/not accepted. This prevents to setup a call.

I took a look to the source code and found that in the file: chan_sip.c no i= handling is present. (May be its something else).

The correct behaviour is: The i= flag and the following text (e.g. i=This is a company call) is set by a Caller and send to a Callee. It must not changed in the asterisk it just has to be forwarded in the SDP file to the Callee.

However using the same call with my local sip provider (sipgate) the packet is accepted and forwarded to the Callee.

An topic according to this issue exists in the forum:

Kind Regards
Comments:By: Leif Madsen (lmadsen) 2011-10-31 14:34:21.317-0500

While this is technically a feature request, I do feel it is a valid request, so I'm acknowledging it. However the priority for this issue is going to be relatively low. I would recommend providing a patch for this functionality in order to move this issue forward. Thanks!

By: Joshua C. Colp (jcolp) 2015-03-14 10:18:53.702-0500

Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch.


[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Additionally if this still remains an issue please provide SIP configuration, SIP traces, and Asterisk logging.