[asterisk-users] PJSIP signaling question

[George Joseph]

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
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