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List: asterisk-dev
Subject: [asterisk-dev] [Code Review] 2894: rtp_engine: ast_rtp_instance_early_bridge_make_compatible confusi
From: "Scott Griepentrog" <reviewboard () asterisk ! org>
Date: 2013-09-30 23:02:27
Message-ID: 20130930230227.10754.49967 () hotblack ! digium ! com
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2894/
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Review request for Asterisk Developers.
Bugs: ASTERISK-21464
https://issues.asterisk.org/jira/browse/ASTERISK-21464
Repository: Asterisk
Description
-------
Testing for Issue 21464 (thanks Kevin!) turned up some odd behavior on codec \
negotiation, including loss of codec payloads. The arguments to \
ast_rtp_instance_early_bridge_make_compatible were not clearly indicating source and \
destination channel for the copy of codecs, which made it non-obvious that the \
arguments to ast_rtp_codecs_payloads_copy() were reversed. The result is that an \
offer containing 119 telephone event would be converted to 101 telephone event, for \
both directrtpsetup=yes and no.
Diffs
-----
/branches/1.8/include/asterisk/rtp_engine.h 400206
/branches/1.8/main/rtp_engine.c 400206
Diff: https://reviewboard.asterisk.org/r/2894/diff/
Testing
-------
Tested with 1.8 to prove that telephone-event payload code 119 is now being passed \
again (as it was in Asterisk versions prior to 1.8).
Thanks,
Scott Griepentrog
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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2894/">https://reviewboard.asterisk.org/r/2894/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Scott Griepentrog.</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21464">ASTERISK-21464</a>
</div>
<div style="margin-top: 1.5em;">
<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>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" \
style="border: 1px solid #b8b5a0"> <tr>
<td>
<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;">Testing for Issue 21464 (thanks Kevin!) turned up some odd behavior on \
codec negotiation, including loss of codec payloads. The arguments to \
ast_rtp_instance_early_bridge_make_compatible were not clearly indicating source and \
destination channel for the copy of codecs, which made it non-obvious that the \
arguments to ast_rtp_codecs_payloads_copy() were reversed. The result is that an \
offer containing 119 telephone event would be converted to 101 telephone event, for \
both directrtpsetup=yes and no.</pre> </td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: \
1px solid #b8b5a0"> <tr>
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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;">Tested with 1.8 to prove that telephone-event payload code 119 is now \
being passed again (as it was in Asterisk versions prior to 1.8).</pre> </td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/branches/1.8/include/asterisk/rtp_engine.h <span style="color: \
grey">(400206)</span></li>
<li>/branches/1.8/main/rtp_engine.c <span style="color: grey">(400206)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/2894/diff/" style="margin-left: \
3em;">View Diff</a></p>
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