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