Audio not received from WhatsApp WebRTC INVITE via external profile #2904
Description
Description
When receiving a SIP/SDES WebRTC INVITE from WhatsApp, FreeSWITCH answers the call, but no audio is audible. Softphone calls work correctly with the same IVR, codecs, and dialplan.
Steps to Reproduce
Configure FreeSWITCH external profile with a public IP.
Receive SIP INVITE from WhatsApp WebRTC endpoint.
Call is answered, but audio from WhatsApp → FreeSWITCH is silent.
Logs show STUN role warnings:
switch_rtp.c:1047 sofia/external/[email protected] got audio stun binding response 487
switch_rtp.c:1055 audio STUN Changing role to CONTROLLING
Current Profile Configuration
SDP Offer from WhatsApp
v=0
o=- 1756796996455 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 615fc575-4123-4434-8cd2-af31a2ecf0e7
a=ice-lite
m=audio 3484 UDP/TLS/RTP/SAVPF 111 126
c=IN IP4 157.240.227.51
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1071287312 1 udp 2122260223 157.240.227.51 3484 typ host generation 0 network-cost 50
a=ice-ufrag:xvHbQrdjBJChfCKq
a=ice-pwd:w76mjojNhCCjdeNRVlrTpQ==
a=fingerprint:sha-256 9A:AF:2D:81:E9:25:B7:68:21:29:F1:52:DA:37:1F:89:68:82:BD:DA:0A:EC:81:14:6A:30:C2:40:47:5B:13:DC
a=setup:actpass
a=mid:audio
a=msid:615fc575-4123-4434-8cd2-af31a2ecf0e7 WhatsAppTrack1
a=rtcp-mux
a=rtcp-fb:111 transport-cc
a=maxptime:20
a=ptime:20
a=ssrc:1185525788 cname:WhatsAppAudioStream1
FreeSWITCH SDP Answer
v=0
o=FreeSWITCH 1756790003 1756790004 IN IP4 15.185.161.128
s=FreeSWITCH
c=IN IP4 15.185.161.128
t=0 0
a=msid-semantic: WMS r43lYlc596dUlFhtQQhYhPg75IaRm7qb
m=audio 17354 UDP/TLS/RTP/SAVPF 111 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=20
a=rtpmap:126 telephone-event/8000
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 D7:AB:09:CE:ED:14:F2:14:1B:0C:1E:51:E6:56:6A:F7:7A:98:6C:E0:24:C9:44:01:29:85:23:51:7D:44:EB:6C
a=setup:active
a=rtcp-mux
a=rtcp:17354 IN IP4 15.185.161.128
a=ice-ufrag:y8Zdc15BPvQQUiRG
a=ice-pwd:fTYey9Tx5VBkZFwWpXaz8h1K
a=candidate:3655409713 1 udp 2130706431 15.185.161.128 17354 typ host generation 0
a=end-of-candidates
Observations
Softphone calls work perfectly with the same IVR.
WhatsApp WebRTC calls: FreeSWITCH answers, but audio is silent.
Logs indicate STUN role conflict (487 binding response).
UDP connectivity tested; ports are reachable.
Expected Behavior
FreeSWITCH should complete ICE/DTLS negotiation and audio should flow from WhatsApp to FreeSWITCH.
Environment
FreeSWITCH version: <YOUR_VERSION>
OS: <YOUR_OS>
Public IP: 15.185.161.128
WebRTC client: WhatsApp Web / WhatsApp API SIP INVITE
Attachments / Logs
SDP offer & answer included above.
STUN logs:
switch_rtp.c:1047 sofia/external/[email protected] got audio stun binding response 487
switch_rtp.c:1055 audio STUN Changing role to CONTROLLING
Additional Notes
Tried both ice-lite=true and without; no change.
Public IP is used directly; firewall allows RTP ports.
Opus codec loaded.
Request
Guidance on FreeSWITCH configuration to correctly handle WhatsApp WebRTC / SDES audio.
Possible bug in handling ICE-lite / DTLS / SDES from WhatsApp.
Labels: bug, WebRTC, ice-lite, external profile, audio