The callee gets the ring tone where the caller doesn’t get the ringtone in webrtc call.
Audio is at 13370
Video is at 10.214.0.18:13530
Adding codec opus to SDP
Adding video codec vp8 to SDP
Adding video codec h264 to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.176.16.131:51675:
INVITE sip:o536qbns@192.0.2.174;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.214.0.18:5060;branch=z9hG4bK079c66f7;rport
Max-Forwards: 70
From: “ERU_CAD2” sip:8738382838@10.214.0.18;tag=as4a938533
To: sip:o536qbns@192.0.2.174;transport=wss
Contact: sip:8738382838@10.214.0.18:5060;transport=ws
Call-ID: 195326ce13e20466302291dc46d6a508@10.214.0.18:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.18.0
Date: Fri, 20 Dec 2024 04:44:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1693v=0
o=root 294862860 294862860 IN IP4 10.214.0.18
s=Asterisk PBX 18.18.0
c=IN IP4 10.214.0.18
b=CT:5120
t=0 0
m=audio 13370 UDP/TLS/RTP/SAVPF 107 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:2eee532c606df9b24bc1940f63c8e718
a=ice-pwd:05436d0d1de541b50ac6c13b3f6f6523
a=candidate:Had60012 1 UDP 2130706431 10.214.0.18 13370 typ host
a=candidate:Se8baddb 1 UDP 1694498815 14.139.173.219 13370 typ srflx raddr 10.214.0.18 rport 13370
a=candidate:Had60012 2 UDP 2130706430 10.214.0.18 13371 typ host
a=candidate:Se8baddb 2 UDP 1694498814 14.139.173.219 13371 typ srflx raddr 10.214.0.18 rport 13371
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 BB:53:6B:AC:3F:F4:C3:52:46:A0:B1:4E:F6:5F:89:66:62:E2:67:D7:0B:82:0B:46:30:D9:1C:8D:F3:F5:C9:E3
a=rtcp-mux
a=sendrecv
m=video 13530 UDP/TLS/RTP/SAVPF 100 99
a=ice-ufrag:7768f0f558e8a4ac612b39085496d061
a=ice-pwd:7bbf33622e09a6772ae1a8465ac3eab6
a=candidate:Had60012 1 UDP 2130706431 10.214.0.18 13530 typ host
a=candidate:Se8baddb 1 UDP 1694498815 14.139.173.219 13530 typ srflx raddr 10.214.0.18 rport 13530
a=candidate:Had60012 2 UDP 2130706430 10.214.0.18 13531 typ host
a=candidate:Se8baddb 2 UDP 1694498814 14.139.173.219 13531 typ srflx raddr 10.214.0.18 rport 13531
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 BB:53:6B:AC:3F:F4:C3:52:46:A0:B1:4E:F6:5F:89:66:62:E2:67:D7:0B:82:0B:46:30:D9:1C:8D:F3:F5:C9:E3
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=rtcp-mux
a=sendrecv
-- Called SIP/201004 > 0x7fd3681f7930 -- Strict RTP switching to RTP target address 10.176.16.223:61898 as source<— SIP read from WS:10.176.16.131:51675 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.214.0.18:5060;branch=z9hG4bK079c66f7;rport
From: “ERU_CAD2” sip:8738382838@10.214.0.18;tag=as4a938533
To: sip:o536qbns@192.0.2.174;transport=wss
CSeq: 102 INVITE
Call-ID: 195326ce13e20466302291dc46d6a508@10.214.0.18:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Content-Length: 0<------------->
— (9 headers 0 lines) —<— SIP read from WS:10.176.16.131:51675 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.214.0.18:5060;branch=z9hG4bK079c66f7;rport
From: “ERU_CAD2” sip:8738382838@10.214.0.18;tag=as4a938533
To: sip:o536qbns@192.0.2.174;transport=wss;tag=n161r0i5rh
CSeq: 102 INVITE
Call-ID: 195326ce13e20466302291dc46d6a508@10.214.0.18:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Contact: sip:o536qbns@192.0.2.174;transport=wss
Content-Length: 0
This log only shows the callee side interactions, so there is no information on what is happening on the caller side.
Resolved by removeing the Answer() application from the dialplan.
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