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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3882/">https://reviewboard.asterisk.org/r/3882/</a>
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<p style="margin-top: 0;">On August 4th, 2014, 1:17 p.m. EDT, <b>Alexander Traud</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3882/diff/1/?file=65918#file65918line3041" style="color: black; font-weight: bold; text-decoration: underline;">trunk/channels/chan_sip.c</a>
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(Diff revision 1)
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<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_tcp_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session,</pre></td>
<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_tcptls_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session,</pre></td>
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<th bgcolor="#e9eaa8" style="border-right: 1px solid #C0C0C0;" align="right"><font size="2">3041</font></th>
<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>if (res < 0) {</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>if (res < 0<span class="hl"> && errno != EAGAIN</span>) {</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;">Thank you for adding me to the list of reviewers. That way, I got E-mail notifications.
No no-go from my (limited set of personal) test-cases, because they all passed successfully.
Except: Case (res < 0 && errno == EAGAIN) is missing which leads to readbuf[-1] = '\0' right now. To prevent this, I went for
if (res < 0) {
if (errno == EAGAIN) {
continue;
} else {
ast_debug(…
return -1;
}
I went for "continue" because that was required to fix an issue with Nokia Symbian/S60 connected over UMTS (SIP INVITE = two TCP messages, 1.3 seconds latency between those messages); an issue introduced by the new variable "exclusive_input" (revision 416071).</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;">Alex -
Thank you for testing and your comments. I have implemented your change as it is more elegant, and added handling for EINTR per rmudgett's advice.</pre>
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<p>- ebroad</p>
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<p>On August 5th, 2014, 10:21 p.m. EDT, ebroad wrote:</p>
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<div>Review request for Asterisk Developers and Alexander Traud.</div>
<div>By ebroad.</div>
<p style="color: grey;"><i>Updated Aug. 5, 2014, 10:21 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-18345">ASTERISK-18345</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;">Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. See https://reviewboard.asterisk.org/r/3653/ for the discussion on this.</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;">Made and received calls successfully with CSipSimple with full SIP headers over TLS, SRTP and multiple codecs enabled ensuring a large SDP payload.</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">(419821)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3882/diff/" style="margin-left: 3em;">View Diff</a></p>
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