I am trying to use a webrtc client with Asterisk. I have tried both SIP.js based client and SIPML5 based client. In both cases my calls connect fine and I have audio in both directions. The issue is that if I hangup the call from my PSTN side, the BYE doesn’t reach the webrtc client on the browser. Although on Asterisk logs it shows it is sending to the correct IP.
Similarly, if I make a call from Asterisk to the WebRTC client, the Invites do not reach the WebRTC client, although on Asterisk it shows they are sent to the correct IP.
Following are my Asterisk console logs:
'''
tts-mimic*CLI>
== WebSocket connection from '152.59.85.212:62964' for protocol 'sip' accepted using version '13'
<--- Received SIP request (481 bytes) from WSS:152.59.85.212:62964 --->
REGISTER sip:phone.nspl.cloud:7987 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK6519341
To: <sip:1988@phone.nspl.cloud:7987>
From: <sip:1988@phone.nspl.cloud:7987>;tag=lui5b10bph
CSeq: 2 REGISTER
Call-ID: 3icuqtbo398i3d1kvaep
Max-Forwards: 70
Contact: <sip:e3sse0f8@dp8gu8f7ln0u.invalid;transport=ws>;expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: SIP.js/0.21.2
Content-Length: 0
<--- Transmitting SIP response (470 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK6519341
Call-ID: 3icuqtbo398i3d1kvaep
From: <sip:1988@phone.nspl.cloud>;tag=lui5b10bph
To: <sip:1988@phone.nspl.cloud>;tag=z9hG4bK6519341
CSeq: 2 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1714034697/9235a2148fc4b77f8c4155b622548d09",opaque="7b7c707b3d463715",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length: 0
<--- Received SIP request (757 bytes) from WSS:152.59.85.212:62964 --->
REGISTER sip:phone.nspl.cloud:7987 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK5569235
To: <sip:1988@phone.nspl.cloud:7987>
From: <sip:1988@phone.nspl.cloud:7987>;tag=lui5b10bph
CSeq: 3 REGISTER
Call-ID: 3icuqtbo398i3d1kvaep
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1988", realm="asterisk", nonce="1714034697/9235a2148fc4b77f8c4155b622548d09", uri="sip:phone.nspl.cloud:7987", response="14286c8f0d5edfc8ee044660e39a672e", opaque="7b7c707b3d463715", qop=auth, cnonce="ll22kr17ljaa", nc=00000001
Contact: <sip:e3sse0f8@dp8gu8f7ln0u.invalid;transport=ws>;expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: SIP.js/0.21.2
Content-Length: 0
-- Added contact 'sip:e3sse0f8@152.59.85.212:62964;transport=ws;x-ast-orig-host=dp8gu8f7ln0u.invalid:0' to AOR '1988' with expiration of 600 seconds
-- Removed contact 'sip:ef1r7fku@152.59.85.212:60581;transport=ws;x-ast-orig-host=844rlc2g9f67.invalid:0' from AOR '1988' due to remove existing
== Contact 1988/sip:ef1r7fku@152.59.85.212:60581;transport=ws;x-ast-orig-host=844rlc2g9f67.invalid:0 has been deleted
<--- Transmitting SIP response (422 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK5569235
Call-ID: 3icuqtbo398i3d1kvaep
From: <sip:1988@phone.nspl.cloud>;tag=lui5b10bph
To: <sip:1988@phone.nspl.cloud>;tag=z9hG4bK5569235
CSeq: 3 REGISTER
Date: Thu, 25 Apr 2024 08:44:57 GMT
Contact: <sip:e3sse0f8@dp8gu8f7ln0u.invalid;transport=ws>;expires=599
Server: Asterisk PBX 20.4.0
Content-Length: 0
<--- Received SIP request (2481 bytes) from WSS:152.59.85.212:62964 --->
INVITE sip:+917777777777@sbc-poc.brokerengage.net:8443 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK1020501
To: <sip:+917777777777@sbc-poc.brokerengage.net:8443>
From: <sip:1988@phone.nspl.cloud:7987>;tag=apsu5jvrpt
CSeq: 1 INVITE
Call-ID: 3icuq9agm1j0qgnb4tvh
Max-Forwards: 70
Contact: <sip:e3sse0f8@dp8gu8f7ln0u.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: 1952
v=0
o=- 6693483440372293490 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 298bfee0-6668-4dd2-bf75-03e725559852
m=audio 64367 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 152.59.85.212
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:762665376 1 udp 2122194687 192.168.31.206 64367 typ host generation 0 network-id 1 network-cost 10
a=candidate:377435172 1 udp 2122262783 2409:40d1:bc:54f5:b467:d1a6:64c1:aec6 55452 typ host generation 0 network-id 2 network-cost 10
a=candidate:1838099209 1 udp 1685987071 152.59.85.212 64367 typ srflx raddr 192.168.31.206 rport 64367 generation 0 network-id 1 network-cost 10
a=candidate:36497837 1 tcp 1518214911 192.168.31.206 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:958825513 1 tcp 1518283007 2409:40d1:bc:54f5:b467:d1a6:64c1:aec6 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=ice-ufrag:hX3Z
a=ice-pwd:B7WFC/lr/ucpUbyk2rlpJN+8
a=ice-options:trickle
a=fingerprint:sha-256 CF:A2:4E:F8:9D:5D:02:92:18:0D:74:CE:1B:6E:60:CF:76:5E:6D:53:C8:3E:89:85:28:7B:02:73:3A:31:89:91
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:298bfee0-6668-4dd2-bf75-03e725559852 b5ba6fb5-ccff-44ea-9216-8970b3b283a2
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:2095178518 cname:4SKANt5Z+4Z7FV7C
a=ssrc:2095178518 msid:298bfee0-6668-4dd2-bf75-03e725559852 b5ba6fb5-ccff-44ea-9216-8970b3b283a2
<--- Transmitting SIP response (485 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK1020501
Call-ID: 3icuq9agm1j0qgnb4tvh
From: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
To: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=z9hG4bK1020501
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1714034704/248b7471e6337f5f414b79cb18b3f761",opaque="3db1539022bcf51f",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length: 0
<--- Received SIP request (329 bytes) from WSS:152.59.85.212:62964 --->
ACK sip:+917777777777@sbc-poc.brokerengage.net:8443 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK1020501
To: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=z9hG4bK1020501
From: <sip:1988@phone.nspl.cloud:7987>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<--- Received SIP request (2778 bytes) from WSS:152.59.85.212:62964 --->
INVITE sip:+917777777777@sbc-poc.brokerengage.net:8443 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK464268
To: <sip:+917777777777@sbc-poc.brokerengage.net:8443>
From: <sip:1988@phone.nspl.cloud:7987>;tag=apsu5jvrpt
CSeq: 2 INVITE
Call-ID: 3icuq9agm1j0qgnb4tvh
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1988", realm="asterisk", nonce="1714034704/248b7471e6337f5f414b79cb18b3f761", uri="sip:+917777777777@sbc-poc.brokerengage.net:8443", response="145b325a6e7d31617cd715fd706e5604", opaque="3db1539022bcf51f", qop=auth, cnonce="a56q75emg6dh", nc=00000001
Contact: <sip:e3sse0f8@dp8gu8f7ln0u.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: 1952
v=0
o=- 6693483440372293490 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 298bfee0-6668-4dd2-bf75-03e725559852
m=audio 64367 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 152.59.85.212
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:762665376 1 udp 2122194687 192.168.31.206 64367 typ host generation 0 network-id 1 network-cost 10
a=candidate:377435172 1 udp 2122262783 2409:40d1:bc:54f5:b467:d1a6:64c1:aec6 55452 typ host generation 0 network-id 2 network-cost 10
a=candidate:1838099209 1 udp 1685987071 152.59.85.212 64367 typ srflx raddr 192.168.31.206 rport 64367 generation 0 network-id 1 network-cost 10
a=candidate:36497837 1 tcp 1518214911 192.168.31.206 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:958825513 1 tcp 1518283007 2409:40d1:bc:54f5:b467:d1a6:64c1:aec6 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=ice-ufrag:hX3Z
a=ice-pwd:B7WFC/lr/ucpUbyk2rlpJN+8
a=ice-options:trickle
a=fingerprint:sha-256 CF:A2:4E:F8:9D:5D:02:92:18:0D:74:CE:1B:6E:60:CF:76:5E:6D:53:C8:3E:89:85:28:7B:02:73:3A:31:89:91
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:298bfee0-6668-4dd2-bf75-03e725559852 b5ba6fb5-ccff-44ea-9216-8970b3b283a2
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:2095178518 cname:4SKANt5Z+4Z7FV7C
a=ssrc:2095178518 msid:298bfee0-6668-4dd2-bf75-03e725559852 b5ba6fb5-ccff-44ea-9216-8970b3b283a2
<--- Transmitting SIP response (313 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK464268
Call-ID: 3icuq9agm1j0qgnb4tvh
From: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
To: <sip:+917777777777@sbc-poc.brokerengage.net>
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Content-Length: 0
-- Executing [+917777777777@dialout:1] Verbose("PJSIP/1988-00000009", "1,Dialing out +917777777777") in new stack
Dialing out +917777777777
-- Executing [+917777777777@dialout:2] Set("PJSIP/1988-00000009", "CALLERID(all)="+917777777777" <+917777777777>") in new stack
-- Executing [+917777777777@dialout:3] MixMonitor("PJSIP/1988-00000009", "/tmp/recording.wav") in new stack
-- Executing [+917777777777@dialout:4] Dial("PJSIP/1988-00000009", "PJSIP/+917777777777@idt") in new stack
== Begin MixMonitor Recording PJSIP/1988-00000009
-- Called PJSIP/+917777777777@idt
<--- Transmitting SIP request (1039 bytes) to UDP:64.64.5.7:5060 --->
INVITE sip:+917777777777@64.64.5.7:5060 SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjc51ea1ff-a92e-4515-86e3-cf1354e28281
From: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
To: <sip:+917777777777@64.64.5.7>
Contact: <sip:asterisk@64.64.56.78:56565>
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
CSeq: 13557 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+917777777777" <sip:+917777777777@64.64.56.78>
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Type: application/sdp
Content-Length: 263
v=0
o=- 1180786355 1180786355 IN IP4 64.64.56.78
s=Asterisk
c=IN IP4 64.64.56.78
t=0 0
m=audio 12834 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (462 bytes) from UDP:64.64.5.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.64.56.78:56565;branch=z9hG4bKPjc51ea1ff-a92e-4515-86e3-cf1354e28281;received=64.64.56.78;rport=56565
From: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
To: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
Contact: <sip:64.64.5.7:5060>
CSeq: 13557 INVITE
Server: sbc_5
Content-Length: 0
<--- Received SIP response (806 bytes) from UDP:64.64.5.7:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 64.64.56.78:56565;branch=z9hG4bKPjc51ea1ff-a92e-4515-86e3-cf1354e28281;received=64.64.56.78;rport=56565
From: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
To: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
Contact: <sip:64.64.5.7:5060>
CSeq: 13557 INVITE
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
Server: sbc_5
Content-Type: application/sdp
Content-Length: 244
v=0
o=SBCSIPUAS 1641474734 1 IN IP4 169.132.219.99
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 169.132.219.99
t=0 0
m=audio 20948 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:101 0-15
a=maxptime:150
> 0x7ff648034d60 -- Strict RTP learning after remote address set to: 169.132.219.99:20948
-- PJSIP/idt-0000000a is making progress passing it to PJSIP/1988-00000009
> 0x7ff644038ee0 -- Strict RTP learning after remote address set to: 152.59.85.212:64367
<--- Transmitting SIP response (1590 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK464268
Call-ID: 3icuq9agm1j0qgnb4tvh
From: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
To: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Contact: <sip:64.64.56.78:56565>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 1045
v=0
o=- 3042549618 4 IN IP4 64.64.56.78
s=Asterisk
c=IN IP4 64.64.56.78
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 14914 UDP/TLS/RTP/SAVPF 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 CD:97:00:CA:35:0C:86:5D:95:2E:D5:D3:F6:19:29:10:82:CF:36:9B:19:80:A4:25:21:AE:07:AF:DC:FE:31:63
a=ice-ufrag:6d14482c49269e4100b22257781ad660
a=ice-pwd:1b19f3a3325f975e43c7eee70a501a8d
a=candidate:H40e3834b 1 UDP 2130706431 64.64.56.78 14914 typ host
a=candidate:Ha2f0005 1 UDP 2130706431 10.47.0.5 14914 typ host
a=candidate:Ha7a0002 1 UDP 2130706431 10.122.0.2 14914 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 14914 typ host
a=candidate:Hac1c0001 1 UDP 2130706431 172.28.0.1 14914 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:1939512802 cname:b5da6f5b-23da-4a2b-821f-1b27876086ea
a=msid:c42d7321-4181-498d-8c1c-ffbc25ce7908 fc0203d1-1399-4e68-a9d1-bbc7beebe659
a=rtcp-fb:* transport-cc
a=mid:0
<--- Received SIP response (792 bytes) from UDP:64.64.5.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.64.56.78:56565;branch=z9hG4bKPjc51ea1ff-a92e-4515-86e3-cf1354e28281;received=64.64.56.78;rport=56565
From: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
To: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
Contact: <sip:64.64.5.7:5060>
CSeq: 13557 INVITE
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
Server: sbc_5
Content-Type: application/sdp
Content-Length: 244
v=0
o=SBCSIPUAS 1641474734 1 IN IP4 169.132.219.99
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 169.132.219.99
t=0 0
m=audio 20948 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:101 0-15
a=maxptime:150
<--- Transmitting SIP request (448 bytes) to UDP:64.64.5.7:5060 --->
ACK sip:64.64.5.7:5060 SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjbe58f264-0d28-4d38-bf1c-0ef6b726bc26
From: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
To: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
CSeq: 13557 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
-- PJSIP/idt-0000000a answered PJSIP/1988-00000009
<--- Transmitting SIP response (1624 bytes) to WSS:152.59.85.212:62964 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;rport=62964;received=152.59.85.212;branch=z9hG4bK464268
Call-ID: 3icuq9agm1j0qgnb4tvh
From: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
To: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:64.64.56.78:56565>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1045
v=0
o=- 3042549618 4 IN IP4 64.64.56.78
s=Asterisk
c=IN IP4 64.64.56.78
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 14914 UDP/TLS/RTP/SAVPF 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 CD:97:00:CA:35:0C:86:5D:95:2E:D5:D3:F6:19:29:10:82:CF:36:9B:19:80:A4:25:21:AE:07:AF:DC:FE:31:63
a=ice-ufrag:6d14482c49269e4100b22257781ad660
a=ice-pwd:1b19f3a3325f975e43c7eee70a501a8d
a=candidate:H40e3834b 1 UDP 2130706431 64.64.56.78 14914 typ host
a=candidate:Ha2f0005 1 UDP 2130706431 10.47.0.5 14914 typ host
a=candidate:Ha7a0002 1 UDP 2130706431 10.122.0.2 14914 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 14914 typ host
a=candidate:Hac1c0001 1 UDP 2130706431 172.28.0.1 14914 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:1939512802 cname:b5da6f5b-23da-4a2b-821f-1b27876086ea
a=msid:c42d7321-4181-498d-8c1c-ffbc25ce7908 fc0203d1-1399-4e68-a9d1-bbc7beebe659
a=rtcp-fb:* transport-cc
a=mid:0
-- Channel PJSIP/idt-0000000a joined 'simple_bridge' basic-bridge <c434f6ef-b1d7-4f3c-a5c7-fb2edf53942f>
-- Channel PJSIP/1988-00000009 joined 'simple_bridge' basic-bridge <c434f6ef-b1d7-4f3c-a5c7-fb2edf53942f>
<--- Received SIP request (380 bytes) from WSS:152.59.85.212:62964 --->
ACK sip:64.64.56.78:56565 SIP/2.0
Via: SIP/2.0/WSS dp8gu8f7ln0u.invalid;branch=z9hG4bK6144392
To: <sip:+917777777777@sbc-poc.brokerengage.net:8443>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
From: <sip:1988@phone.nspl.cloud:7987>;tag=apsu5jvrpt
CSeq: 2 ACK
Call-ID: 3icuq9agm1j0qgnb4tvh
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0
> 0x7ff644038ee0 -- Strict RTP learning after ICE completion
> 0x7ff644038ee0 -- Strict RTP learning after remote address set to: 152.59.85.212:64367
> 0x7ff644038ee0 -- Strict RTP switching to RTP target address 152.59.85.212:64367 as source
> 0x7ff648034d60 -- Strict RTP switching to RTP target address 169.132.219.99:20948 as source
> 0x7ff648034d60 -- Strict RTP learning complete - Locking on source address 169.132.219.99:20948
> 0x7ff644038ee0 -- Strict RTP learning complete - Locking on source address 152.59.85.212:64367
<--- Received SIP request (407 bytes) from UDP:64.64.5.7:5060 --->
BYE sip:asterisk@64.64.56.78:56565 SIP/2.0
Via: SIP/2.0/UDP 64.64.5.7:5060;branch=z9hG4bK-1641474734-2
To: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
From: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
CSeq: 13558 BYE
Max-Forwards: 70
Server: sbc_5
Content-Length: 0
<--- Transmitting SIP response (405 bytes) to UDP:64.64.5.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.64.5.7:5060;rport=5060;received=64.64.5.7;branch=z9hG4bK-1641474734-2
Call-ID: 27fb3ad9-6e03-412b-ac86-5d2a7aead049
From: <sip:+917777777777@64.64.5.7>;tag=sbcsipuas_1_C49105_20240425044504562_sbc13
To: "+917777777777" <sip:+917777777777@64.64.56.78>;tag=c240841f-4226-417a-b859-ec5f98544052
CSeq: 13558 BYE
Server: Asterisk PBX 20.4.0
Content-Length: 0
-- Channel PJSIP/idt-0000000a left 'simple_bridge' basic-bridge <c434f6ef-b1d7-4f3c-a5c7-fb2edf53942f>
-- Channel PJSIP/1988-00000009 left 'simple_bridge' basic-bridge <c434f6ef-b1d7-4f3c-a5c7-fb2edf53942f>
== Spawn extension (dialout, +917777777777, 4) exited non-zero on 'PJSIP/1988-00000009'
== MixMonitor close filestream (mixed)
<--- Transmitting SIP request (445 bytes) to UDP:152.59.85.212:62964 --->
BYE sip:e3sse0f8@152.59.85.212:62964;transport=ws;ob SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjae3b8c81-c36d-4e6f-bced-ba7c6004234f
From: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
To: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 27256 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
== End MixMonitor Recording PJSIP/1988-00000009
<--- Transmitting SIP request (445 bytes) to UDP:152.59.85.212:62964 --->
BYE sip:e3sse0f8@152.59.85.212:62964;transport=ws;ob SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjae3b8c81-c36d-4e6f-bced-ba7c6004234f
From: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
To: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 27256 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
<--- Transmitting SIP request (445 bytes) to UDP:152.59.85.212:62964 --->
BYE sip:e3sse0f8@152.59.85.212:62964;transport=ws;ob SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjae3b8c81-c36d-4e6f-bced-ba7c6004234f
From: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
To: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 27256 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
<--- Transmitting SIP request (445 bytes) to UDP:152.59.85.212:62964 --->
BYE sip:e3sse0f8@152.59.85.212:62964;transport=ws;ob SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjae3b8c81-c36d-4e6f-bced-ba7c6004234f
From: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
To: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 27256 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
<--- Transmitting SIP request (445 bytes) to UDP:152.59.85.212:62964 --->
BYE sip:e3sse0f8@152.59.85.212:62964;transport=ws;ob SIP/2.0
Via: SIP/2.0/UDP 64.64.56.78:56565;rport;branch=z9hG4bKPjae3b8c81-c36d-4e6f-bced-ba7c6004234f
From: <sip:+917777777777@sbc-poc.brokerengage.net>;tag=01a2091e-4669-46b5-8e0e-017eaa01e88a
To: <sip:1988@phone.nspl.cloud>;tag=apsu5jvrpt
Call-ID: 3icuq9agm1j0qgnb4tvh
CSeq: 27256 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length: 0
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