[Freeswitch-users] Changing RTP ports mid call

[Joli Martinez]

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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