[asterisk-users] PJSIP signaling question

[Kevin Long]

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.

Regards,

Kevin Long
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 3587 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/1b51ef0b/attachment.bin>