On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <[email protected]> wrote:
> > Thanks George I appreciate the info . Being able to see what codec is in > use for call in progress is very handy sometimes. > > As far as the RTP stats goes, I see there is some info with “rtp” and > “rtcp” commands which can be useful for troubleshooting. A running tally of > # packets or bandwidth used would be awesome in along with the codec in > "pjsip show channels" or something like that. > > > Im not certain, but I think the TLS signalling problem from this email may > be happening to me again after patching for another pjsip/NAT issue which > was with the external_media_address not working and the internal IP being > sent in the SDP from asterisk - I applied this patch to the codebase and > recompiled I am seeing the TLS “new transport” issue again , I think. > I've lost track of who's applying what patches to which codebase. :) Which patch did you apply for "external_media_address not working"? > > Regards, > > Kevin Long > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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