JsSip - Failed to parse SessionDescription. a=fingerprint:SHA-256 Failed to create fingerprint from the digest

pedrazadixon · December 6, 2019, 5:19pm

I try deploy asterisk with jssip this is my confs files:

sip.conf

[general]
context = default
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = no
progressinband = yes
regextenonqualify = yes
rtcachefriends = yes
rtsavesysname = yes
rtupdate = yes
rtautoclear = 3
qualify = 1000
qualifyfreq=10
canreinvite = no
transport=udp,wss
websocket_enabled = true
disallow = all
allow = ulaw
nat=force_rport,comedia
directmedia=no
externip = mi_public_ip:5060
localnet=mi_public_ip/255.255.255.0
rtcp_mux=yes
tlsdontverifyserver=yes

#include peers.conf

peers.conf

; PhonerLite account
[1001]
accountcode=1001
dial = SIP/1001
secret=mi_pass
host=dynamic
context=default
call-limit=1
qualify=yes
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
transport=udp
nat=force_rport,comedia

; JsSip account
[2001]
accountcode=2001
dial = SIP/2001
secret=mi_pass
host=dynamic
context=default
call-limit=1
qualify=yes
type=friend
dtmfmode=rfc2833
avpf=yes
;force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=opus
allow=ulaw
rtcp_mux=yes
transport=wss
dtlssetup=actpass
nat=force_rport,comedia
encryption=yes
dtlsenable=yes
dtlsverify=fingerprint
tlscertfile=/etc/asterisk/keys/asterisk_ok.pem
tlsprivatekey=/etc/asterisk/keys/asterisk_ok.pem

extensions.conf

[default]
exten => 1001,1,Playback(hello-world)
same => n,Dial(SIP/1001)
same => n,Hangup()

exten => 2001,1,Playback(hello-world)
same => n,Dial(SIP/2001)
same => n,Hangup()

exten => 600,1,Answer()
same => n,Playback(demo-echotest)
same => n,Hangup()

I get this error when try make out or in call to the jssip extension (2001)

JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to parse SessionDescription. a=fingerprint:SHA-256  Failed to create fingerprint from the digest.
    at RTCPeerConnection.<computed> [as setRemoteDescription]

any help for me?

david551 · December 6, 2019, 6:55pm

You would need to provide the SDP line that is objecting to.

pedrazadixon · December 6, 2019, 7:22pm

the error is on the last part of 1st post. (sorry for my english, I not speak this. )

yes, bellow the log when i try call to 600 ext (a playback) from 2001 ext (chrome jssip).

this is jssip (console log):

jssip.js:23458 JsSIP:WebSocketInterface new() [url:"wss://pbx_server_ip:8089/ws"] +0ms
jssip.js:23458 JsSIP:UA new() [configuration:{sockets: Array(1), uri: "sip:2001@pbx_server_ip", password: "******", session_timers: false}] +4ms
jssip.js:23458 JsSIP:Transport new() +2ms
jssip.js:23458 JsSIP:UA configuration parameters after validation: +2ms
jssip.js:23458 JsSIP:UA - authorization_user: "2001" +0ms
jssip.js:23458 JsSIP:UA - password: NOT SHOWN +0ms
jssip.js:23458 JsSIP:UA - realm: null +1ms
jssip.js:23458 JsSIP:UA - ha1: NOT SHOWN +0ms
jssip.js:23458 JsSIP:UA - display_name: null +0ms
jssip.js:23458 JsSIP:UA - uri: sip:2001@pbx_server_ip +0ms
jssip.js:23458 JsSIP:UA - contact_uri: {"_parameters":{"transport":"ws"},"_headers":{},"_scheme":"sip","_user":"ans6pb48","_host":"vt7lv6uf4cb4.invalid","_port":null} +0ms
jssip.js:23458 JsSIP:UA - instance_id: "5e864a37-5f10-469a-94c1-54460548d1d4" +0ms
jssip.js:23458 JsSIP:UA - use_preloaded_route: false +0ms
jssip.js:23458 JsSIP:UA - session_timers: false +0ms
jssip.js:23458 JsSIP:UA - no_answer_timeout: 60000 +1ms
jssip.js:23458 JsSIP:UA - register: true +0ms
jssip.js:23458 JsSIP:UA - register_expires: 600 +0ms
jssip.js:23458 JsSIP:UA - registrar_server: sip:pbx_server_ip +0ms
jssip.js:23458 JsSIP:UA - connection_recovery_max_interval: null +0ms
jssip.js:23458 JsSIP:UA - connection_recovery_min_interval: null +0ms
jssip.js:23458 JsSIP:UA - via_host: "vt7lv6uf4cb4.invalid" +0ms
jssip.js:23458 JsSIP:UA start() +1ms
jssip.js:23458 JsSIP:Transport connect() +0ms
jssip.js:23458 JsSIP:WebSocketInterface connect() +0ms
jssip.js:23458 JsSIP:WebSocketInterface connecting to WebSocket wss://pbx_server_ip:8089/ws +0ms
jssip.js:23458 JsSIP:WebSocketInterface WebSocket wss://pbx_server_ip:8089/ws connected +141ms
jssip.js:23458 JsSIP:Transport send() +3ms
jssip.js:23458 JsSIP:Transport sending message:REGISTER sip:pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4088060
Max-Forwards: 69
To: <sip:2001@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=0f17t4eu7p
Call-ID: kd1f026lefrtrojuu4g297
CSeq: 1 REGISTER
Contact: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:5e864a37-5f10-469a-94c1-54460548d1d4>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +68ms
jssip.js:23458 JsSIP:Transport received text message:SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4088060;received=181.61.114.116;rport=49859
From: <sip:2001@pbx_server_ip>;tag=0f17t4eu7p
To: <sip:2001@pbx_server_ip>;tag=as28cb2b7c
Call-ID: kd1f026lefrtrojuu4g297
CSeq: 1 REGISTER
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c2569ab"
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:DigestAuthentication authenticate() | response generated +8ms
jssip.js:23458 JsSIP:Transport send() +1ms
jssip.js:23458 JsSIP:Transport sending message:REGISTER sip:pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK6704618
Max-Forwards: 69
To: <sip:2001@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=0f17t4eu7p
Call-ID: kd1f026lefrtrojuu4g297
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="2001", realm="asterisk", nonce="5c2569ab", uri="sip:pbx_server_ip", response="ef2844d883363bebfa5d42ee87354ff2"
Contact: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:5e864a37-5f10-469a-94c1-54460548d1d4>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +40ms
jssip.js:23458 JsSIP:Transport received text message:OPTIONS sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS pbx_server_ip:5060;branch=z9hG4bK6ff5b3e3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@pbx_server_ip>;tag=as478d983c
To: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>
Contact: <sip:asterisk@pbx_server_ip:5060;transport=ws>
Call-ID: 73a75a3a3cc66acb5e19a75d32e88eb9@pbx_server_ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 06 Dec 2019 19:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:Transport send() +3ms
jssip.js:23458 JsSIP:Transport sending message:SIP/2.0 200 OK
Via: SIP/2.0/WS pbx_server_ip:5060;branch=z9hG4bK6ff5b3e3;rport
To: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>;tag=esntd2jbhm
From: "asterisk" <sip:asterisk@pbx_server_ip>;tag=as478d983c
Call-ID: 73a75a3a3cc66acb5e19a75d32e88eb9@pbx_server_ip:5060
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Accept: application/sdp, application/dtmf-relay
Supported: outbound
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +0ms
jssip.js:23458 JsSIP:Transport received text message:SIP/2.0 200 OK
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK6704618;received=181.61.114.116;rport=49859
From: <sip:2001@pbx_server_ip>;tag=0f17t4eu7p
To: <sip:2001@pbx_server_ip>;tag=as28cb2b7c
Call-ID: kd1f026lefrtrojuu4g297
CSeq: 2 REGISTER
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>;expires=600
Date: Fri, 06 Dec 2019 19:11:58 GMT
Content-Length: 0

 +1ms
jssip.js:23458 JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bK6ff5b3e3 +1ms
jssip.js:23458 JsSIP:UA call() +8s
jssip.js:23458 JsSIP:RTCSession new +0ms
jssip.js:23458 JsSIP:RTCSession connect() +1ms
jssip.js:23458 JsSIP:RTCSession newRTCSession() +2ms
jssip.js:23458 JsSIP:RTCSession unmute() +0ms
jssip.js:23458 JsSIP:RTCSession unmute() +2ms
jssip.js:23458 JsSIP:RTCSession emit "peerconnection" +72ms
jssip.js:23458 JsSIP:RTCSession session connecting +0ms
jssip.js:23458 JsSIP:RTCSession emit "connecting" +0ms
jssip.js:23458 JsSIP:RTCSession createLocalDescription() +0ms
jssip.js:23458 JsSIP:RTCSession emit "sdp" +105ms
jssip.js:23458 JsSIP:RTCSession emit "sending" [request:InitialOutgoingInviteRequest {ua: UA, headers: {…}, method: "INVITE", ruri: URI, body: "v=0
↵o=- 5915947797024394002 2 IN IP4 127.0.0.1
↵s…9437 label:c2daa853-b28b-487b-b7f1-1ba619aa1a17
↵", …}] +0ms
jssip.js:23458 JsSIP:Transport send() +1ms
jssip.js:23458 JsSIP:Transport sending message:INVITE sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4549751
Max-Forwards: 69
To: <sip:600@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1934 INVITE
Contact: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 1896

v=0
o=- 5915947797024394002 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi
m=audio 61409 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1434301788 1 udp 2122260223 192.168.0.10 61409 typ host generation 0 network-id 1 network-cost 10
a=candidate:469649836 1 tcp 1518280447 192.168.0.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:fTB3
a=ice-pwd:eNjE4lwwFHmCwbtqBOw11uTs
a=ice-options:trickle
a=fingerprint:sha-256 7A:5A:BF:B6:DC:13:BA:E9:49:44:8A:6E:50:8A:0B:5E:04:BC:EF:67:5D:54:F1:16:81:4C:D5:88:65:F2:95:EA
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=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi c2daa853-b28b-487b-b7f1-1ba619aa1a17
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1250149437 cname:/VwZUvIh7VcOj+NW
a=ssrc:1250149437 msid:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi c2daa853-b28b-487b-b7f1-1ba619aa1a17
a=ssrc:1250149437 mslabel:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi
a=ssrc:1250149437 label:c2daa853-b28b-487b-b7f1-1ba619aa1a17
 +0ms
jssip.js:23458 JsSIP:Transport send() +2ms
jssip.js:23458 JsSIP:Transport sending message:ACK sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4549751
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as08ce2ded
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1934 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:DigestAuthentication authenticate() | response generated +1ms
jssip.js:23458 JsSIP:Transport send() +0ms
jssip.js:23458 JsSIP:Transport sending message:INVITE sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK7801742
Max-Forwards: 69
To: <sip:600@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1935 INVITE
Authorization: Digest algorithm=MD5, username="2001", realm="asterisk", nonce="55955302", uri="sip:600@pbx_server_ip", response="5abffe57c472ca8aa0b7e861229a88a5"
Contact: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 1896

v=0
o=- 5915947797024394002 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi
m=audio 61409 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1434301788 1 udp 2122260223 192.168.0.10 61409 typ host generation 0 network-id 1 network-cost 10
a=candidate:469649836 1 tcp 1518280447 192.168.0.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:fTB3
a=ice-pwd:eNjE4lwwFHmCwbtqBOw11uTs
a=ice-options:trickle
a=fingerprint:sha-256 7A:5A:BF:B6:DC:13:BA:E9:49:44:8A:6E:50:8A:0B:5E:04:BC:EF:67:5D:54:F1:16:81:4C:D5:88:65:F2:95:EA
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=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi c2daa853-b28b-487b-b7f1-1ba619aa1a17
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1250149437 cname:/VwZUvIh7VcOj+NW
a=ssrc:1250149437 msid:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi c2daa853-b28b-487b-b7f1-1ba619aa1a17
a=ssrc:1250149437 mslabel:Eqr6EuV90P9V8Lkzo3ZeIyqwmD8E2vk4UBMi
a=ssrc:1250149437 label:c2daa853-b28b-487b-b7f1-1ba619aa1a17
 +1ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:InviteClientTransaction Timer D expired for transaction z9hG4bK4549751 +0ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +174ms
jssip.js:23458 JsSIP:Transport received text message:SIP/2.0 100 Trying
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK7801742;received=181.61.114.116;rport=49859
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
To: <sip:600@pbx_server_ip>
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1935 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:600@pbx_server_ip:5060;transport=ws>
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:RTCSession receiveInviteResponse() +2ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +0ms
jssip.js:23458 JsSIP:Transport received text message:SIP/2.0 200 OK
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK7801742;received=181.61.114.116;rport=49859
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
To: <sip:600@pbx_server_ip>;tag=as236eceaf
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1935 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:600@pbx_server_ip:5060;transport=ws>
Content-Type: application/sdp
Content-Length: 824

v=0
o=root 830410215 830410215 IN IP4 pbx_server_ip
s=Asterisk PBX 15.5.0
c=IN IP4 pbx_server_ip
t=0 0
m=audio 13780 UDP/TLS/RTP/SAVPF 111 0 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=ice-ufrag:4348fafe1df044366e9d9d917fce7ba1
a=ice-pwd:39db5dd611d859ed35b53f7852d4eb47
a=candidate:Hc0a8001f 1 UDP 2130706431 192.168.0.31 13780 typ host
a=candidate:Sb58fac6d 1 UDP 1694498815 pbx_server_ip 13780 typ srflx raddr 192.168.0.31 rport 13780
a=candidate:Hc0a8001f 2 UDP 2130706430 192.168.0.31 13781 typ host
a=candidate:Sb58fac6d 2 UDP 1694498814 pbx_server_ip 13781 typ srflx raddr 192.168.0.31 rport 13781
a=connection:new
a=setup:active
a=fingerprint:SHA-256 
a=rtcp-mux
a=sendrecv
 +0ms
jssip.js:23458 JsSIP:RTCSession receiveInviteResponse() +1ms
jssip.js:23458 JsSIP:Dialog new UAC dialog created with status CONFIRMED +1ms
jssip.js:23458 JsSIP:RTCSession emit "sdp" +0ms
jssip.js:23458 JsSIP:RTCSession acceptAndTerminate() +0ms
jssip.js:23458 JsSIP:RTCSession sendRequest() +1ms
jssip.js:23458 JsSIP:Transport send() +0ms
jssip.js:23458 JsSIP:Transport sending message:ACK sip:600@pbx_server_ip:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK505551
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as236eceaf
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1935 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +1ms
jssip.js:23458 JsSIP:RTCSession sendRequest() +0ms
jssip.js:23458 JsSIP:Transport send() +1ms
jssip.js:23458 JsSIP:Transport sending message:BYE sip:600@pbx_server_ip:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4337477
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as236eceaf
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1936 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:RTCSession session failed +0ms
jssip.js:23458 JsSIP:RTCSession close() +0ms
jssip.js:23458 JsSIP:RTCSession emit "failed" +0ms
jssip.js:23637 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to parse SessionDescription. a=fingerprint:SHA-256  Failed to create fingerprint from the digest.
    at RTCPeerConnection.<computed> [as setRemoteDescription] (https://webrtc-jssip.test/static/js/jssip.js:26068:45)
    at https://webrtc-jssip.test/static/js/jssip.js:16799:49] +1ms
debug @ jssip.js:23637
(anonymous) @ jssip.js:16810
Promise.catch (async)
(anonymous) @ jssip.js:16806
Promise.then (async)
_receiveInviteResponse @ jssip.js:16798
onReceiveResponse @ jssip.js:16577
_receiveResponse @ jssip.js:18733
onReceiveResponse @ jssip.js:18643
receiveResponse @ jssip.js:20133
onTransportData @ jssip.js:21972
_onData @ jssip.js:20922
_onMessage @ jssip.js:22982
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +41ms
jssip.js:23458 JsSIP:Transport received text message:SIP/2.0 200 OK
Via: SIP/2.0/WSS vt7lv6uf4cb4.invalid;branch=z9hG4bK4337477;received=181.61.114.116;rport=49859
From: <sip:2001@pbx_server_ip>;tag=c3mfcsudmi
To: <sip:600@pbx_server_ip>;tag=as236eceaf
Call-ID: mdd4p2mdtff2be4f8nab
CSeq: 1936 BYE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface received WebSocket message +2s
jssip.js:23458 JsSIP:Transport received text message:OPTIONS sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS pbx_server_ip:5060;branch=z9hG4bK54973cdc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@pbx_server_ip>;tag=as55b39a7c
To: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>
Contact: <sip:asterisk@pbx_server_ip:5060;transport=ws>
Call-ID: 6ca635283a4d113f196cef0a509af03e@pbx_server_ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 06 Dec 2019 19:12:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:Transport send() +2ms
jssip.js:23458 JsSIP:Transport sending message:SIP/2.0 200 OK
Via: SIP/2.0/WS pbx_server_ip:5060;branch=z9hG4bK54973cdc;rport
To: <sip:ans6pb48@vt7lv6uf4cb4.invalid;transport=ws>;tag=ub75jff245
From: "asterisk" <sip:asterisk@pbx_server_ip>;tag=as55b39a7c
Call-ID: 6ca635283a4d113f196cef0a509af03e@pbx_server_ip:5060
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Accept: application/sdp, application/dtmf-relay
Supported: outbound
Content-Length: 0

 +0ms
jssip.js:23458 JsSIP:WebSocketInterface send() +0ms
jssip.js:23458 JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bK54973cdc +2ms

and this is the sip log from asterisk:

Retransmitting #9 (NAT) to 46.166.142.108:52254:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 46.166.142.108:52254;branch=z9hG4bK81232826;received=46.166.142.108;rport=52254
From: <sip:Bhabha@pbx_server_ip>;tag=1436817608
To: <sip:00441519470339@pbx_server_ip>;tag=as6f7de3a8
Call-ID: 2056072558-342828044-1157019034
CSeq: 1 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from WS:181.61.114.116:49791 --->
INVITE sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK2138472
Max-Forwards: 69
To: <sip:600@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5722 INVITE
Contact: <sip:8tmo4hjq@hoj4hklgu4nc.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 1896

v=0
o=- 1741699277017176208 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE
m=audio 63741 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1434301788 1 udp 2122260223 192.168.0.10 63741 typ host generation 0 network-id 1 network-cost 10
a=candidate:469649836 1 tcp 1518280447 192.168.0.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:2b6O
a=ice-pwd:DVkZJnm2hUP/2ZQBHGvXAhiY
a=ice-options:trickle
a=fingerprint:sha-256 9A:4E:63:00:CC:D3:61:09:B4:65:A5:FC:55:1D:5A:01:67:BC:00:2C:A2:4A:01:3B:70:A1:37:EC:39:C6:2D:7A
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=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE d329db39-0e99-4673-927e-1230c8371264
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1902904034 cname:Mqu0pq9hCQfa94BO
a=ssrc:1902904034 msid:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE d329db39-0e99-4673-927e-1230c8371264
a=ssrc:1902904034 mslabel:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE
a=ssrc:1902904034 label:d329db39-0e99-4673-927e-1230c8371264
<------------->
--- (13 headers 45 lines) ---
Using INVITE request as basis request - pd2bfbqagc4h81fnvegm
Found peer '2001' for '2001' from 181.61.114.116:49791

<--- Reliably Transmitting (NAT) to 181.61.114.116:49791 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK2138472;received=181.61.114.116;rport=49791
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
To: <sip:600@pbx_server_ip>;tag=as5282a59a
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5722 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69c58415"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'pd2bfbqagc4h81fnvegm' in 6400 ms (Method: INVITE)

<--- SIP read from WS:181.61.114.116:49791 --->
ACK sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK2138472
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as5282a59a
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5722 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from WS:181.61.114.116:49791 --->
INVITE sip:600@pbx_server_ip SIP/2.0
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK5513585
Max-Forwards: 69
To: <sip:600@pbx_server_ip>
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5723 INVITE
Authorization: Digest algorithm=MD5, username="2001", realm="asterisk", nonce="69c58415", uri="sip:600@pbx_server_ip", response="4353191d5efe8fb272711a32a7ef8109"
Contact: <sip:8tmo4hjq@hoj4hklgu4nc.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.0.22
Content-Length: 1896

v=0
o=- 1741699277017176208 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE
m=audio 63741 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1434301788 1 udp 2122260223 192.168.0.10 63741 typ host generation 0 network-id 1 network-cost 10
a=candidate:469649836 1 tcp 1518280447 192.168.0.10 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:2b6O
a=ice-pwd:DVkZJnm2hUP/2ZQBHGvXAhiY
a=ice-options:trickle
a=fingerprint:sha-256 9A:4E:63:00:CC:D3:61:09:B4:65:A5:FC:55:1D:5A:01:67:BC:00:2C:A2:4A:01:3B:70:A1:37:EC:39:C6:2D:7A
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=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE d329db39-0e99-4673-927e-1230c8371264
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1902904034 cname:Mqu0pq9hCQfa94BO
a=ssrc:1902904034 msid:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE d329db39-0e99-4673-927e-1230c8371264
a=ssrc:1902904034 mslabel:9bXXY6dOZiHj0s49YSZYatUiU4ceWM90kkmE
a=ssrc:1902904034 label:d329db39-0e99-4673-927e-1230c8371264
<------------->
--- (14 headers 45 lines) ---
Using INVITE request as basis request - pd2bfbqagc4h81fnvegm
Found peer '2001' for '2001' from 181.61.114.116:49791
  == DTLS ECDH initialized (automatic), faster PFS enabled
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 112
Found unknown media description format telephone-event for ID 113
Found audio description format telephone-event for ID 126
Capabilities: us - (opus|ulaw), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (opus|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
       > 0x7f4b0000b6a0 -- Strict RTP learning after remote address set to: 192.168.0.10:63741
Peer audio RTP is at port 192.168.0.10:63741
Looking for 600 in default (domain pbx_server_ip)
sip_route_dump: route/path hop: <sip:8tmo4hjq@hoj4hklgu4nc.invalid;transport=ws;ob>

<--- Transmitting (NAT) to 181.61.114.116:49791 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK5513585;received=181.61.114.116;rport=49791
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
To: <sip:600@pbx_server_ip>
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5723 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:600@pbx_server_ip:5060;transport=ws>
Content-Length: 0


<------------>
    -- Executing [600@default:1] Answer("SIP/2001-00000003", "") in new stack
Audio is at 13828
Adding codec opus to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 181.61.114.116:49791 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK5513585;received=181.61.114.116;rport=49791
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
To: <sip:600@pbx_server_ip>;tag=as27cc8c16
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5723 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:600@pbx_server_ip:5060;transport=ws>
Content-Type: application/sdp
Content-Length: 826

v=0
o=root 1382031689 1382031689 IN IP4 pbx_server_ip
s=Asterisk PBX 15.5.0
c=IN IP4 pbx_server_ip
t=0 0
m=audio 13828 UDP/TLS/RTP/SAVPF 111 0 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=ice-ufrag:2271db111f2fc8072a9ec73d2ed1eecb
a=ice-pwd:4c17c4245c4a87f65a78a89e302c5ef0
a=candidate:Hc0a8001f 1 UDP 2130706431 192.168.0.31 13828 typ host
a=candidate:Sb58fac6d 1 UDP 1694498815 pbx_server_ip 13828 typ srflx raddr 192.168.0.31 rport 13828
a=candidate:Hc0a8001f 2 UDP 2130706430 192.168.0.31 13829 typ host
a=candidate:Sb58fac6d 2 UDP 1694498814 pbx_server_ip 13829 typ srflx raddr 192.168.0.31 rport 13829
a=connection:new
a=setup:active
a=fingerprint:SHA-256
a=rtcp-mux
a=sendrecv

<------------>

<--- SIP read from WS:181.61.114.116:49791 --->
ACK sip:600@pbx_server_ip:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK9399427
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as27cc8c16
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5723 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from WS:181.61.114.116:49791 --->
BYE sip:600@pbx_server_ip:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK3389910
Max-Forwards: 69
To: <sip:600@pbx_server_ip>;tag=as27cc8c16
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5724 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.22
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog 'pd2bfbqagc4h81fnvegm' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 181.61.114.116:49791 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS hoj4hklgu4nc.invalid;branch=z9hG4bK3389910;received=181.61.114.116;rport=49791
From: <sip:2001@pbx_server_ip>;tag=kn2sklt9hg
To: <sip:600@pbx_server_ip>;tag=as27cc8c16
Call-ID: pd2bfbqagc4h81fnvegm
CSeq: 5724 BYE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

thanks in advance for your response.

david551 · December 6, 2019, 9:48pm

That does look like a syntax violation, by Asterisk, in the response SDP in the 200 OK. The RFC says there must be at least one octet of hex. Assuming you are using the latest sub-version of Asterisk, I think you need to raise an issue. I don’t really know anything about Asterisk WebRTC, so I won’t speculate on the cause.

pedrazadixon · December 6, 2019, 10:39pm

thank you for your answer but I don’t completely understand you.
Neither am I an expert in asterisk is my first time with this type of integrations.
I can’t find the line exactly like you.
Where can I start looking?

david551 · December 6, 2019, 11:05pm

Do a text search.

However, if you really are using WebRTC, the consensus here seems to be that you have to be able read SIP traces in your sleep, as well as have detailed knowledge of several other technologies.

RFC 4572 is what defines the allowable format of this SDP parameter.

jcolp · December 7, 2019, 12:32am

Asterisk 15 is no longer supported, so issues aren’t accepted for it. Upgrading would be needed which may resolve the problem, failing that checking the console for any certificate related messages may be a good idea. As well chan_sip is only community supported, so it is unlikely anyone would look at such a problem soon.

system · January 6, 2020, 12:40am

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