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@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@gmail.com <javascript:_e(%7B%7D,'cvml','colin.morelli@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@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>> wrote:
we already re-open the auto adjust window on reinvite
On Friday, May 13, 2016, Colin Morelli <colin.morelli@gmail.com <javascript:_e(%7B%7D,'cvml','colin.morelli@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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