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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2113/">https://reviewboard.asterisk.org/r/2113/</a>
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<p style="margin-top: 0;">On September 19th, 2012, 1:57 p.m., <b>opticron</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2113/diff/1/?file=31234#file31234line9721" style="color: black; font-weight: bold; text-decoration: underline;">/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(struct sip_pvt *p, struct sip_request *req, int t38action)</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> if (has_media_stream(p, SDP_AUDIO)) {</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> if (has_media_stream(p, SDP_AUDIO)) {</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;">This should be checked before the transport checking so that SRTP_CRYPTO_OFFER_OK does not get set inadvertently. This applies to the video stream processing as well.</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;">Fixed.</pre>
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<p style="margin-top: 0;">On September 19th, 2012, 1:57 p.m., <b>opticron</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2113/diff/1/?file=31234#file31234line10024" style="color: black; font-weight: bold; text-decoration: underline;">/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(struct sip_pvt *p, struct sip_request *req, int t38action)</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> } if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> } else if (process_sdp_a_dtls(value, p, p->trtp)) {</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;">Instead of setting processed to TRUE, this should result in setting offer->type = SDP_UNKNOWN, breaking out of the stream parameter processing loop, and continuing with the next stream definition since encryption is unsupported for text streams.</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;">If the offer contained DTLS-SRTP it would have already skipped this stream when looking at the protocol and declined it, since that protocol is not supported. As for handling DTLS-SRTP attributes I've removed it since it is not needed and could cause insanity with bad packets.</pre>
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<p style="margin-top: 0;">On September 19th, 2012, 1:57 p.m., <b>opticron</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;">If DTLS-SRTP (or normal SRTP) is offered in session-level SDP and a text stream is also offered in the same SDP offer, Asterisk should either reject the text stream or the entire offer since it cannot honor the request.</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;">Above.</pre>
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<p>- Joshua</p>
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<p>On September 14th, 2012, 11:43 a.m., Joshua Colp wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Joshua Colp.</div>
<p style="color: grey;"><i>Updated Sept. 14, 2012, 11:43 a.m.</i></p>
<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;">WebRTC has migrated to using DTLS-SRTP as the method for securing media streams. This patch adds support for it using OpenSSL. DTLS is used between both sides with the keying material for SRTP extracted from that negotiation.</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 various configurations between two Asterisk instances. Rekeying, verification, etc all appear to work. Unfortunately there are very few DTLS-SRTP implementations in the wild so testing against another implementation has not yet occurred.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/configure <span style="color: grey">(UNKNOWN)</span></li>
<li>/trunk/channels/chan_sip.c <span style="color: grey">(373058)</span></li>
<li>/trunk/channels/sip/include/sip.h <span style="color: grey">(373058)</span></li>
<li>/trunk/configs/sip.conf.sample <span style="color: grey">(373058)</span></li>
<li>/trunk/configure.ac <span style="color: grey">(373058)</span></li>
<li>/trunk/include/asterisk/autoconfig.h.in <span style="color: grey">(373058)</span></li>
<li>/trunk/include/asterisk/rtp_engine.h <span style="color: grey">(373058)</span></li>
<li>/trunk/main/rtp_engine.c <span style="color: grey">(373058)</span></li>
<li>/trunk/res/res_rtp_asterisk.c <span style="color: grey">(373058)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2113/diff/" style="margin-left: 3em;">View Diff</a></p>
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