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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3658/">https://reviewboard.asterisk.org/r/3658/</a>
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<p style="margin-top: 0;">On June 24th, 2014, 7:58 p.m. UTC, <b>opticron</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3658/diff/1/?file=59977#file59977line11201" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)</pre></td>
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<th bgcolor="#e9eaa8" style="border-right: 1px solid #C0C0C0;" align="right"><font size="2">11201</font></th>
<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>} else if (sscanf(a, "fmtp: %30u %255<span class="hl">s</span>", &codec, fmtp_string) == 2) {</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>} else if (sscanf(a, "fmtp: %30u %255<span class="hl">[^\t\n]</span>", &codec, fmtp_string) == 2) {</pre></td>
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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;">It appears that this will need to be merged into 12/trunk when this patch goes in.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This patch was taken from 12/trunk, "backported" to 11 (in ampersands, because all I did, was to extract it from the original Opus/VP8 patch and tested it with my AMR code).</pre>
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<p>- Alexander</p>
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<p>On June 20th, 2014, 3:02 p.m. UTC, Alexander Traud wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Alexander Traud.</div>
<p style="color: grey;"><i>Updated June 20, 2014, 3:02 p.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-23916">ASTERISK-23916</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;">In SIP/SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Here, I propose to backport this one-line change into Asterisk 11 LTS.
This diff is based on (an excerpt of) SVN revision 397526, review 2723, and ASTERISK-21981. The change was tested with Nokia mobile phones, which allow more than one media attribute with the format AMR.</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;">11.10</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/11/channels/chan_sip.c <span style="color: grey">(416067)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3658/diff/" style="margin-left: 3em;">View Diff</a></p>
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