asterisk users - Mar 2016 - PJSIP signaling question

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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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
> --
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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
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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
>
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