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 -------------- 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/20160302/52818e55/attachment.bin>
On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long <kevin.long at haloprivacy.com> wrote:> > > Interesting, thanks George. I pulled Asterisk 13 from git and the new > pjproject from the SVN and will test accordingly . >?Yeah, actually you do need Asterisk 13 from git because pjproject deprecated an api in trunk and we only handle that in the current git 13 branch.?> > > 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 . >?Unfortunately, no to both (at least that I'm aware of). I remember looking at the channel stats a long while back and for some reason didn't go any further. I can re-look.> > > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160301/5e9a4200/attachment.html>
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>
On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long at haloprivacy.com> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/49f61496/attachment-0001.html>