[asterisk-users] PJSIP signaling question

[George Joseph]

On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.long at haloprivacy.com>
wrote:

> 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/
>

​Oh, that one, OK.  ​  It should be merged now so if you 'git pull' on 13
now, you should get it.  The transport re-use issue was in pjproject so is
it possible that you're not compiling against the latest trunk?






>
> 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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