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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2728/">https://reviewboard.asterisk.org/r/2728/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">With nitpick mode on, I found a few issues. All minor.</pre>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7354" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_hangup(struct ast_channel *ast)</pre></td>
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<td bgcolor="#ffc5ce" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Why remove the docs? Updating it is better =)</pre>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7360" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_hangup(struct ast_channel *ast)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>struct ast_format_cap *jointcaps_copy;</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Call this original_jointcaps. We see that it gets copied. The reason why is more important.</pre>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7384" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_hangup(struct ast_channel *ast)</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>ast_getformatbyname(codec, &fmt);</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>ast_getformatbyname(codec, &fmt);</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I know this was here before you came, but:
according to the docs, you should check the return value (NULL), not fmt.id.
(And we can drop the ast_format_clear() above.)</pre>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7386" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_hangup(struct ast_channel *ast)</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I know this was here before you came, but:
we read a couple of different variables, but claim that it was "${SIP_CODEC}". Perhaps replace with "${SIP_CODEC*}" to avoid confusing the reader.</pre>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7391" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_hangup(struct ast_channel *ast)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_log(LOG_NOTICE, "Set codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Sounds more like a debug message to me. Especially if we're doing a NOTICE message for every codec added.</pre>
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<p>- wdoekes</p>
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<p>On August 1st, 2013, 12:42 a.m. UTC, Matt Jordan wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Matt Jordan.</div>
<p style="color: grey;"><i>Updated Aug. 1, 2013, 12:42 a.m.</i></p>
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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-21976">ASTERISK-21976</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;">{quote}
For video calls, we would like to set the codecs in the dialplan using
SIP_CODEC. However, if SIP_CODEC is set, all codecs except the ONE set are disallowed and thus either audio or video is available.
Attached is a patch for 11.4 that allows SIP_CODEC to contain a list of codecs , e.g. "gsm,h264".
{quote}
As an aside, chan_pjsip has an analogous dialplan function "PJSIP_MEDIA_OFFER". While this doesn't allow for setting multiple codecs, it does handle multiple media types, as you can specify both video or audio for the codec you want to apply - hence I didn't port this patch/feature over to chan_pjsip. If we think it needs it, it would be reasonably easy to do so.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/trunk/channels/chan_sip.c <span style="color: grey">(395907)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2728/diff/" style="margin-left: 3em;">View Diff</a></p>
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