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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4093/">https://reviewboard.asterisk.org/r/4093/</a>
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<h1 style="margin-right: 0.2em; padding: 0; font-size: 10pt;">This change has been marked as submitted.</h1>
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<div>Review request for Asterisk Developers.</div>
<div>By Frankie Chin.</div>
<p style="color: grey;"><i>Updated Dec. 1, 2014, 12:51 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Committed in revision 428708</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24274">ASTERISK-24274</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Currently SLIN12, SLIN24, SLIN32, SLIN44, SLIN48, SLIN96 and SLIN192 are found not working with SIP. The following error will be thrown if one of those codecs is used: chan_sip.c:10718 process_sdp: No compatible codecs, not accepting this offer!
What I think the issue is that the codec format isn't being included in the SDP media attributes when one of those codecs is used. Please refer to ASTERISK-24274 for more details. This change updates the main/rtp_engine.c and main/frame.c to ensure all these codecs are supported.
Note: SLIN and SLIN16 are working fine.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Specified SLIN48 codec in sip.conf of two Asterisk servers. Used AMI to originate a call from Server A to Server B and then put Server B into a conference hosted in Server A. The above mentioned error was no longer reported and the conference was working as expected.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/tags/12.4.0/main/rtp_engine.c <span style="color: grey">(425756)</span></li>
<li>/tags/12.4.0/main/frame.c <span style="color: grey">(425756)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4093/diff/" style="margin-left: 3em;">View Diff</a></p>
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