On Tue, Jan 15, 2019, at 12:18 PM, Brian J. Murrell wrote: > On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > > How is your endpoint currently configured in asterisk? > > It's configured as a chan_sip peer. > > > Have you tried > > rtp_symmetric to see if the endpoint sends audio to asterisk if > > asterisk > > can send audio back to the client? > > That would require using chan_pjsip wouldn't it? Not that I am opposed > to trying that. I need to use chan_pjsip at some point to be able to > authenticate to my SIP provider for SIP SIMPLE anyway. > > But will rtp_symmetric really solve the problem? Isn't the problem the > setting up of the RTP session, so there is no address and port that it > receives from yet?
The chan_sip module has this implemented under the "nat" option using "comedia" as I recall. It causes media to be sent to where media was originally received from. As for whether it would work or not... this all ultimately depends on how exactly the intermediary behaves, what is let through, what is altered. There's nothing inherent in either chan_sip or chan_pjsip to know/care, as it's just SIP. You'd need to look at the SIP signaling and the SDP to understand what is happening. -- Joshua C. Colp Digium - A Sangoma Company | 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
