[asterisk-bugs] [Asterisk 0012501]: autoframing only works if SDP contains rtpmap

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12501 
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Reported By:                slimey
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   12501
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0-beta7.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-23-2008 04:00 CDT
Last Modified:              05-27-2008 14:26 CDT
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Summary:                    autoframing only works if SDP contains rtpmap
Description: 
I'm currently doing an interop with a SIP provider who requires 10ms
packetisation of media (PCMA Audio). I've set autoframing=yes in [global]
of sip.conf, and "sip show settings" shows that Auto Framing is enabled.

The SDP that comes from the SIP provider looks like:

v=0

o=BroadWorks 8348755 1 IN IP4 172.19.0.20

s=-

c=IN IP4 172.19.0.20

t=0 0

m=audio 24342 RTP/AVP 8

a=ptime:10

However, when Asterisk replies with its own SDP, it's still using 20ms:

v=0

o=BroadWorks 8348747 2 IN IP4 172.19.0.20

s=-

c=IN IP4 172.19.0.20

t=0 0

m=audio 24242 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=fmtp:101 0-11

a=sendrecv

I've checked through chan_sip.c, and from what I can see in the code,
autoframing only works if the incoming SDP has a=rtpmap lines

Around line 6638 (I have a patch of my own in there which may have
affected the line numbers) of chan_sip.c:

                 } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5))
{
                 [...]
                        if (framing && last_rtpmap_codec) {

last_rtpmap_codec is only set when it parses an a=rtpmap line in the SDP

Asterisk should apply the ptime even without an rtpmap line, as rtpmap is
only (as I understand it) _required_ if the m= line references a dynamic
rtp type.
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---------------------------------------------------------------------- 
 svnbot - 05-27-08 14:26  
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 118558

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r118558 | file | 2008-05-27 14:26:29 -0500 (Tue, 27 May 2008) | 4 lines

Fix an issue where codec preferences were not set on dialogs that were not
authenticated via a user or peer and allow framing to work without rtpmap
in the SDP.
(closes issue http://bugs.digium.com/view.php?id=12501)
Reported by: slimey

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http://svn.digium.com/view/asterisk?view=rev&revision=118558 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-27-08 14:26  svnbot         Note Added: 0087391                          
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