[asterisk-users] PJSIP signaling question

[Kevin Long]

Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject from the SVN and will test accordingly .



I have a few more questions about PJSIP in Asterisk 13:


1.  Is there any way to list current ongoing calls and see what codecs are being used in the RTP streams?  With chan_sip,  “sip show channels” did this.  

2. Also with a PJSIP initiated call, is there a way to see how man RTP packets have been sent and received for the call , I am debugging some intermittent 1-way and no-way audio on calls , and I am having trouble figuring out fi it is the client, firewall, or Asterisk/pjsip that is the culprit .


Regards,

Kevin Long
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