Nice! As you noticed (and I confirm it) WebRTC gets BYEs now.
Probably I changed something good and it worked. It’s the same config I posted earlier.
So,
WebRTC: no inbound sound (but there are BYEs)
MicroSIP: no BYEs (but there’s sound)
I keep struggling to make WebRTC work. I hope somebody would advise .. I describe the actual issue in the second half of the post. It is very strange.
Earlier I used jsSIP as a web client library, and as I wrote I could send voice with it, but could not hear sound back from Asterisk.
Somehow jsSIP could not handle ICE candidates well enough and all requests had my local ip address behind a WiFi router, c=IN IP4 192.168/16
I haven’t found a way to configure it and make it work.
At some point I tried the demo site with sipML5 and I got sound back from Asterisk. How happy I was! And hang-ups work both ways.
And I got the proper public IP in requests, c=IN IP4 62.216.59.42
(of course I had to use the special commands from earlier)
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
sipML5 is very slow, it works with candidates for 30 seconds and there’re no good examples or documentation to make it work.
So I tried SIP.JS, and again, with some effort, I got sound from Asterisk and proper address c=IN IP4 62.216.59.42.
But, there’s an opposite issue with both SIP.js and sipML5.
They play sound from Asterisk, always, but often do not sent sound to it.
I have two endpoints, one connects WebRTC with the automatic answer “demo-congrats” and the other connects WebRTC with an actual phone. Both endpoints have identical configuration in pjsip.
Ground truth is the MixMonitor in Asterisk which records all calls.
Calls usually work like this.
(50% time) I call “demo-congrats” and I hear sound and I say something and hang up and I can hear both automatic voice and my voice in the recording. I can repeat it several times, all works. RTP packets both ways.
res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:50255 (via ICE) (type 00, seq 017579, ts 675587032, len 000170)
res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:50255 (type 00, seq 023167, ts 805573497, len 000170)
I reload pjsip. Just “pjsip reload” without any changes.
(50% time) I call “demo-congrats” and say something and in the recording there’s only automatic voice, no voice from me. RTP packets one way, towards WebRTC client.
res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:63281 (via ICE) (type 111, seq 060556, ts 181536, len 000126)
res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:63281 (via ICE) (type 111, seq 060557, ts 182496, len 000137)
res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:63281 (via ICE) (type 111, seq 060558, ts 183456, len 000141)
I reload pjsip and can repeat it and get sound both ways. It works again. Sometimes it does not.
Or sometimes it breaks in the middle of a call. Only 1-2 seconds of my voice is recorded, then the WebRTC client becomes “muted” and sound goes only from Asterisk side.
It works the same on both clients sipML (demo site) and SIP.js.
(99% time) If I switch the extension to connect WebRTC with an actual phone, most of the time sound goes one way only, no sound from WebRTC side.
1% of the time the sound goes through for 1 second and mutes.
Somehow the extension with the automatic “demo congrats” response works much-much better than a call to external number through a SIP provider.
So, to summarize,
jsSIP can send sound to Asterisk 100% time but can’t get through the transparent router (wifi or mobile provider) to receive sound;
sipML and Sip.JS are better in establishing connection and receive sound 100% time but cannot reliably send sound towards Asterisk. And most probably Asterisk won’t send it to external mobile recipient.
Any ideas?
P.S. I’ve setup a second clean Ubuntu instance with a clean new Asterisk 22 and the same simplistic configuration, and it works the same.
So I guess anybody who wants to try WebRTC and Asterisk would get into this mess.
WebRTC won’t adjust the c=, but it should send an ICE line with the public address.
Not sure I understand, how can I check it?
Here’s a piece of log from a successful two-way call with “demo-congrats”, if it would be enough
[Mar 26 02:03:27] VERBOSE[312877] res_pjsip_logger.c: <--- Received SIP request (2544 bytes) from WSS:62.216.59.42:50324 --->
INVITE sip:380685978789@mypbx.ua SIP/2.0
Via: SIP/2.0/WSS 4sjubg757rp9.invalid;branch=z9hG4bK5763115
To: <sip:380685978789@mypbx.ua>
From: <sip:1060@mypbx.ua>;tag=6pqglr8fo3
CSeq: 2 INVITE
Call-ID: c1rema45tp9ai7fnfk9q
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1060", realm="asterisk", nonce="1742947407/7580f2d56b8c62a4c0b8a56713359310", uri="sip:380685978789@mypbx.ua", response="a4e98eae6752e011dd16c9d4461dc4bb", opaque="039a02a939da1122", qop=auth, cnonce="lkc0m7ode96i", nc=00000001
Contact: <sip:v60khgrp@4sjubg757rp9.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Type: application/sdp
Content-Length: 1768
v=0
o=mozilla...THIS_IS_SDPARTA-99.0 4566192597130638222 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 9C:42:C0:C6:47:25:71:72:CF:FC:F4:B9:3F:D1:63:77:57:9F:CD:5F:E6:6E:94:76:ED:C4:EC:41:CA:CA:76:35
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 54840 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 62.216.59.42
a=candidate:0 1 UDP 2122252543 192.168.56.1 54839 typ host
a=candidate:2 1 UDP 2122187007 192.168.50.217 54840 typ host
a=candidate:4 1 TCP 2105524479 192.168.56.1 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 192.168.50.217 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.56.1 54841 typ host
a=candidate:2 2 UDP 2122187006 192.168.50.217 54842 typ host
a=candidate:4 2 TCP 2105524478 192.168.56.1 9 typ host tcptype active
a=candidate:5 2 TCP 2105458942 192.168.50.217 9 typ host tcptype active
a=candidate:3 1 UDP 1685987327 62.216.59.42 54840 typ srflx raddr 192.168.50.217 rport 54840
a=candidate:3 2 UDP 1685987326 62.216.59.42 54842 typ srflx raddr 192.168.50.217 rport 54842
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:a2bee8a8af6db9214036aff808cfcb18
a=ice-ufrag:86ec6b33
a=mid:0
a=msid:{3c8788e6-1826-45b7-aa25-23327b6da6d3} {d7e8b2ac-21fe-4378-a6d9-e445e45ec578}
a=rtcp:54842 IN IP4 62.216.59.42
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:1940474017 cname:{c05708df-703a-4d70-bb14-23a8fbd37cbd}
[Mar 26 02:03:27] VERBOSE[312877] res_pjsip_logger.c: <--- Transmitting SIP response (302 bytes) to WSS:62.216.59.42:50324 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 4sjubg757rp9.invalid;rport=50324;received=62.216.59.42;branch=z9hG4bK5763115
Call-ID: c1rema45tp9ai7fnfk9q
From: <sip:1060@mypbx.ua>;tag=6pqglr8fo3
To: <sip:380685978789@mypbx.ua>
CSeq: 2 INVITE
Server: Asterisk PBX 22.2.0
Content-Length: 0
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] pbx.c: Executing [380685978789@rtc60:1] MixMonitor("PJSIP/1060-000000c0", "/home/ubuntu/20250326-020327_1060_380685978789.wav") in new stack
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] pbx.c: Executing [380685978789@rtc60:2] Playback("PJSIP/1060-000000c0", "demo-congrats") in new stack
[Mar 26 02:03:27] VERBOSE[312938][C-00000084] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/1060-000000c0
[Mar 26 02:03:27] VERBOSE[312877] res_rtp_asterisk.c: 0x7ce098318d50 -- Strict RTP learning after remote address set to: 62.216.59.42:54840
[Mar 26 02:03:27] VERBOSE[312877] res_pjsip_logger.c: <--- Transmitting SIP response (1523 bytes) to WSS:62.216.59.42:50324 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 4sjubg757rp9.invalid;rport=50324;received=62.216.59.42;branch=z9hG4bK5763115
Call-ID: c1rema45tp9ai7fnfk9q
From: <sip:1060@mypbx.ua>;tag=6pqglr8fo3
To: <sip:380685978789@mypbx.ua>;tag=c7546f37-35b0-46ce-aadb-7a3326e4fbcb
CSeq: 2 INVITE
Server: Asterisk PBX 22.2.0
Contact: <sip:193.169.240.153:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 935
v=0
o=- 4566192597130638222 2 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 29326 UDP/TLS/RTP/SAVPF 109 0 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 41:AC:CE:29:20:79:3B:F0:35:A8:CE:A0:AE:A3:2D:30:7C:2D:13:20:B8:2C:98:44:56:FA:CF:88:93:AB:8C:67
a=ice-ufrag:6204787a4207c5cd3b2d918665f7585b
a=ice-pwd:3557d70d09d78be025ea57d01c341d11
a=candidate:Hc1a9f099 1 UDP 2130706431 193.169.240.153 29326 typ host
a=candidate:Hd9acdfb7 1 UDP 2130706431 fe80::216:3eff:fee9:fa83 29326 typ host
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:973295942 cname:207ea99e-e24a-4a5f-b65f-0380feb8e7bb
a=msid:07b88f05-8b28-44b9-b28d-fae09a0f9357 bc1e8f88-0890-45a4-ad64-750259bb2e23
a=rtcp-fb:* transport-cc
a=mid:0
[Mar 26 02:03:27] VERBOSE[312877] res_pjsip_logger.c: <--- Received SIP request (373 bytes) from WSS:62.216.59.42:50324 --->
ACK sip:193.169.240.153:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 4sjubg757rp9.invalid;branch=z9hG4bK9726554
To: <sip:380685978789@mypbx.ua>;tag=c7546f37-35b0-46ce-aadb-7a3326e4fbcb
From: <sip:1060@mypbx.ua>;tag=6pqglr8fo3
CSeq: 2 ACK
Call-ID: c1rema45tp9ai7fnfk9q
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: 0x7ce098318d50 -- Strict RTP learning after ICE completion
[Mar 26 02:03:27] VERBOSE[224945] res_rtp_asterisk.c: 0x7ce098318d50 -- Strict RTP learning after remote address set to: 62.216.59.42:54840
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:54840 (via ICE) (type 109, seq 045732, ts 000000, len 000109)
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] file.c: <PJSIP/1060-000000c0> Playing 'demo-congrats.gsm' (language 'en')
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: 0x7ce098318d50 -- Strict RTP switching to RTP target address 62.216.59.42:54840 as source
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:54840 (type 109, seq 022712, ts 987633577, len 000066)
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:54840 (type 109, seq 022713, ts 987634537, len 000050)
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:54840 (type 109, seq 022714, ts 987635497, len 000050)
[Mar 26 02:03:27] VERBOSE[312937][C-00000084] res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:54840 (via ICE) (type 109, seq 045733, ts 000960, len 000108)
Everything seems to be working now.
I got sound both ways somehow (the setup is the same, described earlier in this thread, it just miraculously worked at some point).
I had another issue missing ringback tone, fixed it by adding “earlyMedia: true” to SIP.js configuration.