[Freeswitch-users] Changing RTP ports mid call

[Valli A. Vallimamod]

Hi,

In your case freeswitch starts sending the rtp to the advertised ip/port from SDP (124.123.104.212:10488) but detects that the rtp from the phone is coming from another port (9658) so it switches to this port as it the one actually opened by the nat router. You can disable this feature with the channel variable disable_rtp_auto_adjust=true.

Also the "BUG" in the log line is related to freeswitch media bug, not an actual issue.


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 14 Apr 2020, at 18:20, Joli Martinez <mrjoli021 at gmail.com> wrote:
> 
> Hello,
> 
> We have a phone app that ties back to our freeswitch servers.  Recently the android phones are getting one-way audio.  It works on iPhones and desktop clients.  Based on the pcap's and fs_cli I see that the audio ports changed after the sdp negotiations.  Not sure why this is happening also see that the codec is set to "BUG Codec PCMU:0"
> 
> Any Suggestions?
> 
> 2020-04-14 15:56:07.661344 [DEBUG] switch_ivr_async.c:1500 No silence detection configured; assuming start of speech
> 2020-04-14 15:56:07.701439 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0
> 2020-04-14 15:56:07.701439 [DEBUG] switch_rtp.c:1887 rtcp_stats_init: audio ssrc[1321291818] base_seq[30216]
> 2020-04-14 15:56:07.881347 [INFO] switch_rtp.c:7231 Auto Changing audio port from 124.123.104.212:10488 to 124.123.104.212:9658
> 
> Thanks,
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