No audio when calling via webRTC

osm · April 4, 2023, 6:06am

Hello! No audio when I call via webRTC.

SIP logs:


Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Transport | Sending WebSocket message:

INVITE sip:1001@sip.example.com SIP/2.0

Via: SIP/2.0/WSS gjor8jlbs210.invalid;branch=z9hG4bK9355350

To: <sip:1001@sip.example.com>

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

CSeq: 2 INVITE

Call-ID: hmurivr45qnmmdlgpfut

Max-Forwards: 70

Authorization: Digest algorithm=MD5, username="1000", realm="asterisk", nonce="1680523147/d5b566e21d6da96a1c0ac6607e481b9c", uri="sip:1001@sip.example.com", response="d1f0a4800165d17c7f2b02c541db916f", opaque="4b6701971e005e37", qop=auth, cnonce="etht7npq2bvk", nc=00000001

Contact: <sip:gaf4g9jq@gjor8jlbs210.invalid;transport=ws;ob>

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound

User-Agent: SIP.js/0.21.1

Content-Type: application/sdp

Content-Length: 1879

v=0

o=- 2680209688069743287 2 IN IP4 XX.XX.XX.XX

s=-

t=0 0

a=group:BUNDLE 0

a=extmap-allow-mixed

a=msid-semantic: WMS 0af52b6a-1ba8-4df2-a553-c3b3189754f0

m=audio 62708 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

c=IN IP4 XX.XX.XX.XX

a=rtcp:9 IN IP4 0.0.0.0

a=candidate:3316440647 1 udp 2122260223 XX.XX.XX.XX 62707 typ host generation 0 network-id 1

a=candidate:998679279 1 udp 2122194687 XX.XX.XX.XX 62708 typ host generation 0 network-id 2 network-cost 10

a=candidate:3143843039 1 tcp 1518280447 XX.XX.XX.XX 9 typ host tcptype active generation 0 network-id 1

a=candidate:1162428535 1 tcp 1518214911 XX.XX.XX.XX 9 typ host tcptype active generation 0 network-id 2 network-cost 10

a=candidate:1179857115 1 udp 1685987071 XX.XX.XX.XX 62708 typ srflx raddr XX.XX.XX.XX rport 62708 generation 0 network-id 2 network-cost 10

a=ice-ufrag:w3bC

a=ice-pwd:cia+OYAkO3ZyP0D02pzMhtjZ

a=ice-options:trickle

a=fingerprint:sha-256 EA:FC:97:56:B3:F9:F4:5F:9F:CC:90:EC:B4:45:67:3B:38:C9:51:78:17:C4:E7:DA:D4:A3:96:4C:AE:5B:C3:38

a=setup:actpass

a=mid:0

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01

a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid

a=sendrecv

a=msid:0af52b6a-1ba8-4df2-a553-c3b3189754f0 22d5b50f-a65a-49eb-86cd-1a7d1b406971

a=rtcp-mux

a=rtpmap:111 opus/48000/2

a=rtcp-fb:111 transport-cc

a=fmtp:111 minptime=10;useinbandfec=1

a=rtpmap:63 red/48000/2

a=fmtp:63 111/111

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:13 CN/8000

a=rtpmap:110 telephone-event/48000

a=rtpmap:126 telephone-event/8000

a=ssrc:1182888223 cname:VkSa4FYT2/Z+bsly

a=ssrc:1182888223 msid:0af52b6a-1ba8-4df2-a553-c3b3189754f0 22d5b50f-a65a-49eb-86cd-1a7d1b406971

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

SIP/2.0 100 Trying

Via: SIP/2.0/WSS gjor8jlbs210.invalid;rport=60963;received=XX.XX.XX.XX;branch=z9hG4bK9355350

Call-ID: hmurivr45qnmmdlgpfut

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

To: <sip:1001@sip.example.com>

CSeq: 2 INVITE

Server: FPBX-16.0.40(18.16.0)

Content-Length: 0

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Inviter | Inviter.onTrying

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

SIP/2.0 183 Session Progress

Via: SIP/2.0/WSS gjor8jlbs210.invalid;rport=60963;received=XX.XX.XX.XX;branch=z9hG4bK9355350

Call-ID: hmurivr45qnmmdlgpfut

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

To: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

CSeq: 2 INVITE

Server: FPBX-16.0.40(18.16.0)

Contact: <sip:XX.XX.XX.XX:8089;transport=ws>

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Content-Type: application/sdp

Content-Length: 187

v=0

o=- 3438617271 4 IN IP4 XX.XX.XX.XX

s=Asterisk

c=IN IP4 XX.XX.XX.XX

t=0 0

a=group:BUNDLE 0

m=audio 0 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

c=IN IP4 XX.XX.XX.XX

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.invite-dialog | INVITE dialog hmurivr45qnmmdlgpfutpdtd6cfb8j79781ec0-73d4-4e24-a73f-7dbdc7188be9 constructed

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Inviter | Inviter.onProgress

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

SIP/2.0 183 Session Progress

Via: SIP/2.0/WSS gjor8jlbs210.invalid;rport=60963;received=XX.XX.XX.XX;branch=z9hG4bK9355350

Call-ID: hmurivr45qnmmdlgpfut

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

To: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

CSeq: 2 INVITE

Server: FPBX-16.0.40(18.16.0)

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Contact: <sip:XX.XX.XX.XX:8089;transport=ws>

P-Asserted-Identity: "1001(Доступен)" <sip:1001@sip.example.com>

Content-Type: application/sdp

Content-Length: 187

v=0

o=- 3438617271 4 IN IP4 XX.XX.XX.XX

s=Asterisk

c=IN IP4 XX.XX.XX.XX

t=0 0

a=group:BUNDLE 0

m=audio 0 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

c=IN IP4 XX.XX.XX.XX

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Inviter | Inviter.onProgress

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

SIP/2.0 183 Session Progress

Via: SIP/2.0/WSS gjor8jlbs210.invalid;rport=60963;received=XX.XX.XX.XX;branch=z9hG4bK9355350

Call-ID: hmurivr45qnmmdlgpfut

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

To: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

CSeq: 2 INVITE

Server: FPBX-16.0.40(18.16.0)

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Contact: <sip:XX.XX.XX.XX:8089;transport=ws>

P-Asserted-Identity: "1001(Доступен)" <sip:1001@sip.example.com>

Content-Type: application/sdp

Content-Length: 187

v=0

o=- 3438617271 4 IN IP4 XX.XX.XX.XX

s=Asterisk

c=IN IP4 XX.XX.XX.XX

t=0 0

a=group:BUNDLE 0

m=audio 0 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

c=IN IP4 XX.XX.XX.XX

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:07 GMT+0600 (Киргизия) | sip.Inviter | Inviter.onProgress

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:12 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS gjor8jlbs210.invalid;rport=60963;received=XX.XX.XX.XX;branch=z9hG4bK9355350

Call-ID: hmurivr45qnmmdlgpfut

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

To: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

CSeq: 2 INVITE

Server: FPBX-16.0.40(18.16.0)

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Contact: <sip:XX.XX.XX.XX:8089;transport=ws>

Supported: 100rel, timer, replaces, norefersub

P-Asserted-Identity: "1001(Доступен)" <sip:1001@sip.example.com>

Content-Type: application/sdp

Content-Length: 187

v=0

o=- 3438617271 4 IN IP4 XX.XX.XX.XX

s=Asterisk

c=IN IP4 XX.XX.XX.XX

t=0 0

a=group:BUNDLE 0

m=audio 0 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

c=IN IP4 XX.XX.XX.XX

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:12 GMT+0600 (Киргизия) | sip.Inviter | Inviter.onAccept

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:12 GMT+0600 (Киргизия) | sip.invite-dialog | INVITE dialog hmurivr45qnmmdlgpfutpdtd6cfb8j79781ec0-73d4-4e24-a73f-7dbdc7188be9 sending ACK request

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:12 GMT+0600 (Киргизия) | sip.Transport | Sending WebSocket message:

ACK sip:XX.XX.XX.XX:8089;transport=ws SIP/2.0

Via: SIP/2.0/WSS gjor8jlbs210.invalid;branch=z9hG4bK5496279

To: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

From: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

CSeq: 2 ACK

Call-ID: hmurivr45qnmmdlgpfut

Max-Forwards: 70

Supported: outbound

User-Agent: SIP.js/0.21.1

Content-Length: 0

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:12 GMT+0600 (Киргизия) | sip.Inviter | Session hmurivr45qnmmdlgpfutpdtd6cfb8j transitioned to state Established

rollbar.umd.min.js:1 Mon Apr 03 2023 17:59:18 GMT+0600 (Киргизия) | sip.Transport | Received WebSocket text message:

BYE sip:gaf4g9jq@XX.XX.XX.XX:60963;transport=ws;ob SIP/2.0

Via: SIP/2.0/WSS XX.XX.XX.XX:8089;rport;branch=z9hG4bKPjbfdd061d-32c4-474f-8684-3dd52c0ad5fd;alias

From: <sip:1001@sip.example.com>;tag=79781ec0-73d4-4e24-a73f-7dbdc7188be9

To: <sip:1000@sip.example.com>;tag=pdtd6cfb8j

Call-ID: hmurivr45qnmmdlgpfut

CSeq: 18037 BYE

Reason: Q.850;cause=16

Max-Forwards: 70

User-Agent: FPBX-16.0.40(18.16.0)

Content-Length: 0

Any help would be appreciated. Thanks

jcolp · April 4, 2023, 9:44am

You need to examine the fundamentals of WebRTC to see and understand where it is falling apart:

ICE and DTLS primarily in this case.

david551 · April 4, 2023, 10:17am

Also, the screenshot appears to be from the FreePBX GUI, which is not supported here.

osm · April 4, 2023, 10:21am

well, I’ll edit topic

system · May 4, 2023, 10:21am

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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No audio when calling via webRTC

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