[Freeswitch-users] Update RTP Address On Re-Invite

[Michael Jerris]

in may be only on media address change.  You'd be looking on
switch_core_media.c to find it

On Thursday, July 7, 2016, Colin Morelli <colin.morelli at gmail.com> wrote:

> Hey all,
>
> Reviving this thread just to see if anyone has thoughts here?
>
> I've tried digging through the source and can't seem to find where the
> re-opening of auto_adjust happens during a re-invite. As an aside bug (I'll
> report in JIRA) - it looks like the always_auto_adjust RTP bug results in
> very choppy audio.
>
> That said, I don't need always_auto_adjust, as I'm fine with sending a
> re-invite when I need this to happen. It just doesn't seem to be working.
> Is there something else that needs to be set or is this also a bug?
>
> Best,
> Colin
>
> On Fri, May 13, 2016 at 9:34 PM Colin Morelli <colin.morelli at gmail.com
> <javascript:_e(%7B%7D,'cvml','colin.morelli at gmail.com');>> wrote:
>
>> Is this the case even when the RTP IP in the SDP doesn't change? I'm
>> seeing successful re-invites being processed with no auto adjust happening
>> afterwards. I see:
>>
>> 2016-05-13 21:28:24.216684 [DEBUG] sofia.c:7614 Processing updated SDP
>>
>> Indicating that FS did receive the SDP in the re-invite but nothing else
>> about RTP auto adjust afterwards. This is on FreeSWITCH Version
>> 1.6.7-14-d38d065~64bit (-14-d38d065 64bit) running on Debian 8 x86_64.
>>
>> Thanks for the response.
>>
>> Best,
>> Colin
>>
>> On Fri, May 13, 2016 at 9:14 PM Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> we already re-open the auto adjust window on reinvite
>>>
>>> On Friday, May 13, 2016, Colin Morelli <colin.morelli at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','colin.morelli at gmail.com');>> wrote:
>>>
>>>> I have mobile applications running behind NAT. When their reachability
>>>> changes (and the device's local route updates), I want to automatically
>>>> switch the RTP stream to the new address/port combination.
>>>>
>>>> I've tried using the RTP ALWAYS_AUTO_ADJUST bug, but that results in
>>>> *very* choppy audio. I also don't really need it, as the only time I
>>>> care to perform RTP auto-adjust is after an invite session. For this case I
>>>> can safely assume that the client *will* send a re-invite when its
>>>> address changes. Is there any way to perform RTP audio adjust only on a
>>>> re-invite?
>>>>
>>>> Using STUN doesn't seem right for two reasons: 1) on its own it can't
>>>> solve this problem (even if it determines the external IP, the port is
>>>> still wrong). 2) it results in issues on FS "Invalid STUN/ICE packet
>>>> received 20 bytes"
>>>>
>>>> Would appreciate any help.
>>>>
>>>> Best,
>>>> Colin
>>>>
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