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 <[email protected]> 
> wrote:
> 
> 
> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <[email protected]> 
>> 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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