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<body class='hmmessage'><div dir='ltr'><br> <BR><div>> Date: Wed, 6 Jan 2016 08:22:34 -0400<br>> From: jcolp@digium.com<br>> To: asterisk-dev@lists.digium.com<br>> Subject: Re: [asterisk-dev] RTP/SAVP & TLS<br>> <br>> Ross Beer wrote:<br>> > Hi Dev,<br>> ><br>> > In Asterisk 1.8 Snom phones accept calls when RTP/SAVP is set to<br>> > 'mandatory' which means that the RTP/SAVP options appear in the SDP 'm'<br>> > lines. However in Asterisk 13 chan_pjsip, no such lines exist when using<br>> > 'SDES' encryption.<br>> <br>> The "media_encryption=sdes" option turns on SRTP support and thus makes <br>> the media RTP/SAVP. You can also turn on optimistic SRTP support as well <br>> using "media_encryption_optimistic=yes" which will use RTP/AVP but <br>> include a crypto line. I just checked the testsuite tests for SDP <br>> offer/answer and they are passing, I also manually enabled it and <br>> confirmed it is RTP/SAVP. You may have a configuration error.</div><div> </div><div>Snom devices work correctly when 'media_encryption_optimistic=no', when this is set to yes the RTP/SAVP is replaced:</div><div> </div><div>Set to No = "m=audio 41988 RTP/SAVP 8 0 3 101"</div><div> </div><div>Set to Yes = "m=audio 36240 RTP/AVP 8 0 3 101"</div><div> </div><div>I have updated my configuration to not use the optimistic setting.</div><div><br>> <br>> ><br>> > Therefore Snom phones require this option to be set to 'off'. Should<br>> > Asterisk 13 be offering RTP/SAVP in the same way as chan_sip did?<br>> ><br>> > With regards to TLS, devices reject calls if a 'transport=transport-tls'<br>> > is specified. Is this also a bug as it appears that Asterisk doesn't<br>> > re-use an active connection in this situation?<br>> <br>> This is a bug in PJSIP which has an issue on our side[1]. If an explicit <br>> transport is specified PJSIP will not reuse a connection.<br>> <br>> [1] https://issues.asterisk.org/jira/browse/ASTERISK-22658<br>> </div><div> </div><div>Great, I can work around this until a fix is in place.</div><div> </div><div>Thank you for your assistance.</div><div><br>> -- <br>> Joshua Colp<br>> Digium, Inc. | Senior Software Developer<br>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>> Check us out at: www.digium.com & www.asterisk.org<br>> <br>> <br>> -- <br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-dev<br></div> </div></body>
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