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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2827/">https://reviewboard.asterisk.org/r/2827/</a>
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<p style="margin-top: 0;">On September 6th, 2013, 8:07 a.m. CEST, <b>Olle E Johansson</b> wrote:</p>
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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;">This doesn't work well with PRACK or UPDATES. Asterisk could already have gotten an SDP in another message. (I have a branch with PRACK support that is in production). We need to check if we already have gotten an SDP before we decide to hang up the call. I also need to check the RFCs for this situation.</pre>
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<p>On September 6th, 2013, 10:26 a.m. CEST, <b>wdoekes</b> wrote:</p>
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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;">.. or re-INVITES.
I could imagine that some lazy clients skip the SDP in session refreshers.</pre>
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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;">Yes, but if this is a 200 OK it would mean that we would have sent an INVITE without SDP. Will that ever happen? Curious.</pre>
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<p>- Olle E</p>
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<p>On September 5th, 2013, 9:31 p.m. CEST, jrose wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp, Matt Jordan, and Mark Michelson.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated Sept. 5, 2013, 9:31 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-22424">ASTERISK-22424</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;">One of our SIP tests was previously pushing 200 OKs without SDP and Asterisk would accept these calls without question. According to Mark this should not be accepted because there will be no way to know where to send media to or receive media from in these circumstances. The approach this patch takes is to forcibly hang up the call at this point if there is no SDP on the response provided that it's not a response to a reinvite (in which case the behavior is the same as if there were an SDP that couldn't be parsed properly).</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;">Tested it against SIP_hold before and after
Tested it against a number of testsuite tests against SIP (any of the ones I could run before the patch)
Tested regular SIP phone calls (they didn't hit the modified code path though).</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/1.8/channels/chan_sip.c <span style="color: grey">(398378)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2827/diff/" style="margin-left: 3em;">View Diff</a></p>
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