[asterisk-users] PJSIP signaling question

[Kevin Long]

Hi George the patch was from here , you wrote it I believe . I pulled asterisk 13 from git, apply this patch which fixed RTP issue , but I think tla transport issue came back for me . 

https://gerrit.asterisk.org/#/c/2346/

Thank you

Sent from my iPhone

> On Mar 4, 2016, at 12:01 AM, George Joseph <george.joseph at fairview5.com> wrote:
> 
> 
> 
>> 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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