WebRTC & Asterisk 11.7.0 - not hear audio

Asterisk Asterisk Support
1

Hi, I have problem. I configure my asterisk with WebRPC. I use Asterisk 11.7.0 and tryit.jssip.net.

Connection is OK but I’am not hearing audio (playback and conversation)

please, help me :wink:

sip.conf

... [100] ;webrpc type=friend secret=100 host=dynamic context=michal transport=udp,ws,wss avpf=yes encryption=yes icesupport=yes disallow=all allow=alaw allow=ulaw

extensions.conf:

...
[michal]
exten => _X.,1,Playback(tt-monkeys) 
exten => _X.,1,Dial(SIP/${EXTEN})[/code]

http.conf:
[code]...
bindaddr=0.0.0.0
enabled=yes
bindport=8088[/code]


CLI:
[quote]Connected to Asterisk 11.7.0 currently running on vbilling (pid = 3899)
vbilling*CLI> sip reload all
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Using SIP CoS mark 4
  == Parsing '/etc/asterisk/sip_notify.conf': Found
  == Using SIP RTP CoS mark 5
    -- Executing [111@michal:1] Playback("SIP/100-00000003", "tt-monkeys") in new stack
    -- <SIP/100-00000003> Playing 'tt-monkeys.gsm' (language 'en')
vbilling*CLI>[/quote]

[b][color=#FF0000]it is playback but i not heard sound ;([/color][/b]

RTP debug:
[code]vbilling*CLI> rtp set debug on
RTP Debugging Enabled
  == Using SIP RTP CoS mark 5
    -- Executing [1111@michal:1] Playback("SIP/100-00000004", "tt-monkeys") in new stack
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031213, ts 000160, len 000164)
    -- <SIP/100-00000004> Playing 'tt-monkeys.gsm' (language 'en')
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031214, ts 000320, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031215, ts 000480, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031216, ts 000640, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031217, ts 000800, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031218, ts 000960, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031219, ts 001120, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031220, ts 001280, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031221, ts 001440, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031222, ts 001600, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031223, ts 001760, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031224, ts 001920, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031225, ts 002080, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031226, ts 002240, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031227, ts 002400, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031228, ts 002560, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031229, ts 002720, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031230, ts 002880, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031231, ts 003040, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031232, ts 003200, len 000164)
Sent RTP packet to      95.160.158.151:51326 (type 08, seq 031233, ts 003360, len 000164)
vbilling*CLI>
2

Things to check:

  1. The correct IP negotiation in the SDP of both Asterisk and JsSIP

  2. Validate that your Asterisk is compiled with the uuid-devel/libuuid-devel libraries in order to enable ICE(your rtp doesnt show the label “via ICE”).

  3. If you are in the same LAN as the PBX check that the RTP audio goes to the LAN IP not to the Public IP, use the workaround in the JsSIP PI setting the STUN server to ‘null’.

  4. Provide both sip debug logs from JsSIP and Asterisk.

3

@navaismo, thanks for reply

My laptop (client) with Google Chrome - 192.168.51.3
My Asterisk - 192.168.51.9

How check it is?

Its my debug:

[code]JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783
· via_host: “peiululf5p51.invalid” jssip-devel.js:5794
· password: NOT SHOWN jssip-devel.js:5791
· register_expires: 600 jssip-devel.js:5794
· register_min_expires: 120 jssip-devel.js:5794
· register: true jssip-devel.js:5794
· registrar_server: sip:192.168.51.9 jssip-devel.js:5788
· ws_server_max_reconnection: 3 jssip-devel.js:5794
· ws_server_reconnection_timeout: 4 jssip-devel.js:5794
· connection_recovery_min_interval: 2 jssip-devel.js:5794
· connection_recovery_max_interval: 30 jssip-devel.js:5794
· use_preloaded_route: false jssip-devel.js:5794
· no_answer_timeout: 60000 jssip-devel.js:5794
· stun_servers: [“stun:stun.l.google.com:19302”] jssip-devel.js:5794
· turn_servers: jssip-devel.js:5794
· trace_sip: true jssip-devel.js:5794
· hack_via_tcp: false jssip-devel.js:5794
· hack_ip_in_contact: false jssip-devel.js:5794
· uri: sip:100@192.168.51.9 jssip-devel.js:5788
· ws_servers: [{“ws_uri”:“ws://192.168.51.9:8088/ws”,“sip_uri”:“sip:192.168.51.9:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794
· display_name: “100” jssip-devel.js:5794
· instance_id: “7caae24a-d71b-468f-a7c5-0d09c41667db” jssip-devel.js:5794
· jssip_id: “c4j7i” jssip-devel.js:5794
· hostport_params: “192.168.51.9” jssip-devel.js:5794
· authorization_user: “100” jssip-devel.js:5794
JsSIP | EVENT EMITTER | adding event newMessage jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event newRTCSession jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event registrationFailed jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event unregistered jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event registered jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event disconnected jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event connected jssip-devel.js:67
JsSIP | EVENT EMITTER | new listener added to event connected jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event disconnected jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newMessage jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event registered jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event unregistered jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-devel.js:63
JsSIP | UA | user requested startup… jssip-devel.js:5238
JsSIP | TRANSPORT | connecting to WebSocket ws://192.168.51.9:8088/ws jssip-devel.js:568
JsSIP | TRANSPORT | WebSocket ws://192.168.51.9:8088/ws connected jssip-devel.js:604
JsSIP | UA | connection state set to 0 jssip-devel.js:5360
JsSIP | EVENT EMITTER | emitting event connected jssip-devel.js:187
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK7503800
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 81 REGISTER
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:7caae24a-d71b-468f-a7c5-0d09c41667db”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK7503800;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
To: sip:100@192.168.51.9;tag=as51e206b9
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 81 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3d66d490”
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6637211
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“3d66d490”, uri=“sip:192.168.51.9”, response=“2930f180b852ac955bf64740125db4bb”
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:7caae24a-d71b-468f-a7c5-0d09c41667db”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6637211;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
To: sip:100@192.168.51.9;tag=as51e206b9
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 82 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;expires=600
Date: Sun, 12 Jan 2014 20:28:19 GMT
Content-Length: 0

jssip-devel.js:686
JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187
Registered init.js:425
JsSIP | EVENT EMITTER | adding event newDTMF jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event ended jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event started jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event failed jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event progress jssip-devel.js:67
JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-devel.js:187
JsSIP | RTC SESSION | requesting access to local media jssip-devel.js:3442
JsSIP | EVENT EMITTER | new listener added to event progress jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event started jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event failed jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event ended jssip-devel.js:63
JsSIP | RTC SESSION | got local media stream jssip-devel.js:3446
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 INVITE
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3276

v=0
o=- 6843435261791628335 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
m=audio 50270 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2465071814 cname:WDkgccFsgfpOoGcI
a=ssrc:2465071814 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
a=ssrc:2465071814 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:2465071814 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
m=video 50270 RTP/SAVPF 100 116 117
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1336778605 cname:WDkgccFsgfpOoGcI
a=ssrc:1336778605 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0
a=ssrc:1336778605 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:1336778605 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9;tag=as06fb6d4e
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5fbb46dd”
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657
To: sip:100@192.168.51.9;tag=as06fb6d4e
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 ACK

jssip-devel.js:519
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“5fbb46dd”, uri="sip:100@192.168.51.9", response=“044d12332d9e0c16f44e6aa9999d651f”
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3276

v=0
o=- 6843435261791628335 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
m=audio 50270 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2465071814 cname:WDkgccFsgfpOoGcI
a=ssrc:2465071814 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
a=ssrc:2465071814 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:2465071814 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
m=video 50270 RTP/SAVPF 100 116 117
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1336778605 cname:WDkgccFsgfpOoGcI
a=ssrc:1336778605 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0
a=ssrc:1336778605 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:1336778605 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0

jssip-devel.js:519
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK6014657 jssip-devel.js:1969
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.51.9:5060;transport=WS
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9;tag=as77392278
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.51.9:5060;transport=WS
Content-Type: application/sdp
Content-Length: 403

v=0
o=root 703073431 703073431 IN IP4 192.168.51.9
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.51.9
t=0 0
m=audio 11092 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:6m4OuyQDFTxime0a9uRrpL8kSskPEyyzZZJ9v/fJ
m=video 0 RTP/SAVPF 100 116 117

jssip-devel.js:686
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-devel.js:2546
JsSIP | RTC SESSION | stream added: default jssip-devel.js:3392
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:100@192.168.51.9:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK3482275
Max-Forwards: 69
To: sip:100@192.168.51.9;tag=as77392278
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | EVENT EMITTER | emitting event started jssip-devel.js:187
JsSIP | TRANSPORT | received WebSocket text message:

BYE sip:duu35udr@peiululf5p51.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.51.9:5060;branch=z9hG4bK3df6ce15
Max-Forwards: 70
From: sip:100@192.168.51.9;tag=as77392278
To: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.7.0
Proxy-Authorization: Digest username=“r8oumeb5”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.51.9”, nonce=“5fbb46dd”, response=“f746171f1fe50942d851a894252db743”
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.51.9:5060;branch=z9hG4bK3df6ce15
To: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
From: sip:100@192.168.51.9;tag=as77392278
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 102 BYE
Content-Length: 0

jssip-devel.js:519
JsSIP | RTC SESSION | closing INVITE session c4j7imnqn2u69938mpb66d4c8jqn3k jssip-devel.js:4248
JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424
JsSIP | DIALOG | dialog c4j7imnqn2u69938mpb66d4c8jqn3kas77392278 deleted jssip-devel.js:2566
JsSIP | EVENT EMITTER | emitting event ended jssip-devel.js:187
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK3df6ce15 jssip-devel.js:2094
JsSIP | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK5050190 jssip-devel.js:1960
JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK5050190 jssip-devel.js:1949[/code]

4

[quote=“wacky”]@navaismo, thanks for reply

My laptop (client) with Google Chrome - 192.168.51.3
My Asterisk - 192.168.51.9
[/quote]
So this is LAN to LAN, your first RTP debug show a Public IP, you need to fix that. Check your Asterisk peer’s settings about NAT and in the JsSIP API check the ICE or use the workaorund, setting ICE server to null.

Install the libraries, re run the configure script and recompile asterisk.

[quote]
Its my debug:

[code]JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783
· via_host: “peiululf5p51.invalid” jssip-devel.js:5794
· password: NOT SHOWN jssip-devel.js:5791
· register_expires: 600 jssip-devel.js:5794
· register_min_expires: 120 jssip-devel.js:5794
· register: true jssip-devel.js:5794
· registrar_server: sip:192.168.51.9 jssip-devel.js:5788
· ws_server_max_reconnection: 3 jssip-devel.js:5794
· ws_server_reconnection_timeout: 4 jssip-devel.js:5794
· connection_recovery_min_interval: 2 jssip-devel.js:5794
· connection_recovery_max_interval: 30 jssip-devel.js:5794
· use_preloaded_route: false jssip-devel.js:5794
· no_answer_timeout: 60000 jssip-devel.js:5794
· stun_servers: [“stun:stun.l.google.com:19302”] jssip-devel.js:5794
· turn_servers: jssip-devel.js:5794
· trace_sip: true jssip-devel.js:5794
· hack_via_tcp: false jssip-devel.js:5794
· hack_ip_in_contact: false jssip-devel.js:5794
· uri: sip:100@192.168.51.9 jssip-devel.js:5788
· ws_servers: [{“ws_uri”:“ws://192.168.51.9:8088/ws”,“sip_uri”:“sip:192.168.51.9:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794
· display_name: “100” jssip-devel.js:5794
· instance_id: “7caae24a-d71b-468f-a7c5-0d09c41667db” jssip-devel.js:5794
· jssip_id: “c4j7i” jssip-devel.js:5794
· hostport_params: “192.168.51.9” jssip-devel.js:5794
· authorization_user: “100” jssip-devel.js:5794
JsSIP | EVENT EMITTER | adding event newMessage jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event newRTCSession jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event registrationFailed jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event unregistered jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event registered jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event disconnected jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event connected jssip-devel.js:67
JsSIP | EVENT EMITTER | new listener added to event connected jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event disconnected jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newMessage jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event registered jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event unregistered jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-devel.js:63
JsSIP | UA | user requested startup… jssip-devel.js:5238
JsSIP | TRANSPORT | connecting to WebSocket ws://192.168.51.9:8088/ws jssip-devel.js:568
JsSIP | TRANSPORT | WebSocket ws://192.168.51.9:8088/ws connected jssip-devel.js:604
JsSIP | UA | connection state set to 0 jssip-devel.js:5360
JsSIP | EVENT EMITTER | emitting event connected jssip-devel.js:187
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK7503800
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 81 REGISTER
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:7caae24a-d71b-468f-a7c5-0d09c41667db”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK7503800;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
To: sip:100@192.168.51.9;tag=as51e206b9
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 81 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3d66d490”
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6637211
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“3d66d490”, uri=“sip:192.168.51.9”, response=“2930f180b852ac955bf64740125db4bb”
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:7caae24a-d71b-468f-a7c5-0d09c41667db”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6637211;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=95kc5hfjkn
To: sip:100@192.168.51.9;tag=as51e206b9
Call-ID: ushk79hn4ettqgpopuuj2q
CSeq: 82 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;expires=600
Date: Sun, 12 Jan 2014 20:28:19 GMT
Content-Length: 0

jssip-devel.js:686
JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187
Registered init.js:425
JsSIP | EVENT EMITTER | adding event newDTMF jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event ended jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event started jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event failed jssip-devel.js:67
JsSIP | EVENT EMITTER | adding event progress jssip-devel.js:67
JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-devel.js:187
JsSIP | RTC SESSION | requesting access to local media jssip-devel.js:3442
JsSIP | EVENT EMITTER | new listener added to event progress jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event started jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event failed jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-devel.js:63
JsSIP | EVENT EMITTER | new listener added to event ended jssip-devel.js:63
JsSIP | RTC SESSION | got local media stream jssip-devel.js:3446
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
jssip-devel.js:3401
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 INVITE
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3276

v=0
o=- 6843435261791628335 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
m=audio 50270 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2465071814 cname:WDkgccFsgfpOoGcI
a=ssrc:2465071814 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
a=ssrc:2465071814 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:2465071814 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
m=video 50270 RTP/SAVPF 100 116 117
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1336778605 cname:WDkgccFsgfpOoGcI
a=ssrc:1336778605 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0
a=ssrc:1336778605 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:1336778605 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0

jssip-devel.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9;tag=as06fb6d4e
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5fbb46dd”
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK6014657
To: sip:100@192.168.51.9;tag=as06fb6d4e
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2142 ACK

jssip-devel.js:519
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:100@192.168.51.9 SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190
Max-Forwards: 69
To: sip:100@192.168.51.9
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“5fbb46dd”, uri="sip:100@192.168.51.9", response=“044d12332d9e0c16f44e6aa9999d651f”
Contact: sip:duu35udr@peiululf5p51.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3276

v=0
o=- 6843435261791628335 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
m=audio 50270 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2465071814 cname:WDkgccFsgfpOoGcI
a=ssrc:2465071814 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
a=ssrc:2465071814 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:2465071814 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHa0
m=video 50270 RTP/SAVPF 100 116 117
c=IN IP4 95.160.158.151
a=rtcp:50270 IN IP4 95.160.158.151
a=candidate:96503048 1 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:96503048 2 udp 2113937151 192.168.51.3 50270 typ host generation 0
a=candidate:1262713336 1 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:1262713336 2 tcp 1509957375 192.168.51.3 0 typ host generation 0
a=candidate:4090559708 1 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=candidate:4090559708 2 udp 1845501695 95.160.158.151 50270 typ srflx raddr 192.168.51.3 rport 50270 generation 0
a=ice-ufrag:1YUrMq93Rc3A1O1d
a=ice-pwd:3IKU0LhB6+rJfnVtys+cpfCp
a=ice-options:google-ice
a=fingerprint:sha-256 AA:AB:A9:E2:B9:20:71:19:8B:12:67:18:11:30:B8:CD:3E:EE:80:80:99:0C:54:FC:7B:61:4D:75:66:CE:9A:AA
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=sendrecv
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:AQieWl3A0ztd+fcOU2egvVmvtL8vemnFnzSQ82fD
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1336778605 cname:WDkgccFsgfpOoGcI
a=ssrc:1336778605 msid:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0
a=ssrc:1336778605 mslabel:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyH
a=ssrc:1336778605 label:KHzXoqxtMagGHpP2uvhv4mPqot1Fk9e4wgyHv0

jssip-devel.js:519
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK6014657 jssip-devel.js:1969
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.51.9:5060;transport=WS
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK5050190;received=192.168.51.3
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
To: sip:100@192.168.51.9;tag=as77392278
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:100@192.168.51.9:5060;transport=WS
Content-Type: application/sdp
Content-Length: 403

v=0
o=root 703073431 703073431 IN IP4 192.168.51.9
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.51.9
t=0 0
m=audio 11092 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:6m4OuyQDFTxime0a9uRrpL8kSskPEyyzZZJ9v/fJ
m=video 0 RTP/SAVPF 100 116 117

jssip-devel.js:686
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-devel.js:2546
JsSIP | RTC SESSION | stream added: default jssip-devel.js:3392
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:100@192.168.51.9:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS peiululf5p51.invalid;branch=z9hG4bK3482275
Max-Forwards: 69
To: sip:100@192.168.51.9;tag=as77392278
From: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 2143 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | EVENT EMITTER | emitting event started jssip-devel.js:187
JsSIP | TRANSPORT | received WebSocket text message:

BYE sip:duu35udr@peiululf5p51.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.51.9:5060;branch=z9hG4bK3df6ce15
Max-Forwards: 70
From: sip:100@192.168.51.9;tag=as77392278
To: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.7.0
Proxy-Authorization: Digest username=“r8oumeb5”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.51.9”, nonce=“5fbb46dd”, response=“f746171f1fe50942d851a894252db743”
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

jssip-devel.js:686
JsSIP | TRANSPORT | sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.51.9:5060;branch=z9hG4bK3df6ce15
To: “100” sip:100@192.168.51.9;tag=6d4c8jqn3k
From: sip:100@192.168.51.9;tag=as77392278
Call-ID: c4j7imnqn2u69938mpb6
CSeq: 102 BYE
Content-Length: 0

jssip-devel.js:519
JsSIP | RTC SESSION | closing INVITE session c4j7imnqn2u69938mpb66d4c8jqn3k jssip-devel.js:4248
JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424
JsSIP | DIALOG | dialog c4j7imnqn2u69938mpb66d4c8jqn3kas77392278 deleted jssip-devel.js:2566
JsSIP | EVENT EMITTER | emitting event ended jssip-devel.js:187
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK3df6ce15 jssip-devel.js:2094
JsSIP | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK5050190 jssip-devel.js:1960
JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK5050190 jssip-devel.js:1949[/code][/quote]

You debug show a Public IP and you need to provide both logs, so try first fixing the NAT/ICE stuff.

5

@navaismo: i not heard sound ;(

My laptop (client) with Google Chrome - 192.168.0.100
My Asterisk - 192.168.0.101

My debug (only private ip without stun server)

[code]JsSIP | UA | configuration parameters after validation: jssip-0.3.0.js:5733
· via_host: “fhmojhigdgmh.invalid” jssip-0.3.0.js:5741
· password: “100” jssip-0.3.0.js:5741
· register_expires: 600 jssip-0.3.0.js:5741
· register_min_expires: 120 jssip-0.3.0.js:5741
· register: true jssip-0.3.0.js:5741
· registrar_server: sip:192.168.0.101 jssip-0.3.0.js:5738
· ws_server_max_reconnection: 3 jssip-0.3.0.js:5741
· ws_server_reconnection_timeout: 4 jssip-0.3.0.js:5741
· connection_recovery_min_interval: 2 jssip-0.3.0.js:5741
· connection_recovery_max_interval: 30 jssip-0.3.0.js:5741
· use_preloaded_route: false jssip-0.3.0.js:5741
· no_answer_timeout: 60000 jssip-0.3.0.js:5741
· stun_servers: [] jssip-0.3.0.js:5741
· turn_servers: [] jssip-0.3.0.js:5741
· trace_sip: true jssip-0.3.0.js:5741
· hack_via_tcp: false jssip-0.3.0.js:5741
· hack_ip_in_contact: false jssip-0.3.0.js:5741
· uri: sip:100@192.168.0.101 jssip-0.3.0.js:5738
· ws_servers: [{“ws_uri”:“ws://192.168.0.101:8088/ws”,“sip_uri”:“sip:192.168.0.101:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-0.3.0.js:5741
· display_name: “100” jssip-0.3.0.js:5741
· instance_id: “19e4d3d6-48e1-4070-a69c-14496e3ff2fc” jssip-0.3.0.js:5741
· jssip_id: “pnqo0” jssip-0.3.0.js:5741
· hostport_params: “192.168.0.101” jssip-0.3.0.js:5741
· authorization_user: “100” jssip-0.3.0.js:5741
JsSIP | EVENT EMITTER | adding event newMessage jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event newRTCSession jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event registrationFailed jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event unregistered jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event registered jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event disconnected jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event connected jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | new listener added to event connected jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event disconnected jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event newMessage jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event registered jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event unregistered jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-0.3.0.js:63
JsSIP | UA | user requested startup… jssip-0.3.0.js:5180
JsSIP | TRANSPORT | connecting to WebSocket ws://192.168.0.101:8088/ws jssip-0.3.0.js:555
JsSIP | TRANSPORT | WebSocket ws://192.168.0.101:8088/ws connected jssip-0.3.0.js:591
JsSIP | UA | connection state set to 0 jssip-0.3.0.js:5301
JsSIP | EVENT EMITTER | emitting event connected jssip-0.3.0.js:187
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.0.101 SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK877034
Max-Forwards: 69
To: sip:100@192.168.0.101
From: “100” sip:100@192.168.0.101;tag=dmp7331cgj
Call-ID: s44t6gtfv8dg5tjvs8prr0
CSeq: 81 REGISTER
Contact: sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:19e4d3d6-48e1-4070-a69c-14496e3ff2fc”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK877034;received=192.168.0.100
From: “100” sip:100@192.168.0.101;tag=dmp7331cgj
To: sip:100@192.168.0.101;tag=as6d783daf
Call-ID: s44t6gtfv8dg5tjvs8prr0
CSeq: 81 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="63932ab9"
Content-Length: 0

jssip-0.3.0.js:670
JsSIP | TRANSPORT | sending WebSocket message:

REGISTER sip:192.168.0.101 SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK7550784
Max-Forwards: 69
To: sip:100@192.168.0.101
From: “100” sip:100@192.168.0.101;tag=dmp7331cgj
Call-ID: s44t6gtfv8dg5tjvs8prr0
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“63932ab9”, uri=“sip:192.168.0.101”, response="0a1b936a5d5800ccdf6d809529b46ecd"
Contact: sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:19e4d3d6-48e1-4070-a69c-14496e3ff2fc”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK7550784;received=192.168.0.100
From: “100” sip:100@192.168.0.101;tag=dmp7331cgj
To: sip:100@192.168.0.101;tag=as6d783daf
Call-ID: s44t6gtfv8dg5tjvs8prr0
CSeq: 82 REGISTER
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;expires=600
Date: Mon, 13 Jan 2014 08:48:22 GMT
Content-Length: 0

jssip-0.3.0.js:670
JsSIP | EVENT EMITTER | emitting event registered jssip-0.3.0.js:187
Registered init.js:433
JsSIP | EVENT EMITTER | adding event newDTMF jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event ended jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event started jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event failed jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | adding event progress jssip-0.3.0.js:67
JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-0.3.0.js:187
JsSIP | EVENT EMITTER | new listener added to event progress jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event started jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event failed jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-0.3.0.js:63
JsSIP | EVENT EMITTER | new listener added to event ended jssip-0.3.0.js:63
JsSIP | RTC SESSION | requesting access to local media jssip-0.3.0.js:3410
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2131708102 1 udp 2113937151 192.168.0.100 56399 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2131708102 2 udp 2113937151 192.168.0.100 56399 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151 192.168.56.1 56400 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151 192.168.56.1 56400 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1054674330 1 udp 2113937151 192.168.15.174 56401 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1054674330 2 udp 2113937151 192.168.15.174 56401 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:831304758 1 tcp 1509957375 192.168.0.100 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:831304758 2 tcp 1509957375 192.168.0.100 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1885270378 1 tcp 1509957375 192.168.15.174 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1885270378 2 tcp 1509957375 192.168.15.174 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:1111@192.168.0.101 SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK2611438
Max-Forwards: 69
To: sip:1111@192.168.0.101
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5946 INVITE
Contact: sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 2248

v=0
o=- 1211472537477211476 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0
m=audio 56399 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.0.100
a=rtcp:56399 IN IP4 192.168.0.100
a=candidate:2131708102 1 udp 2113937151 192.168.0.100 56399 typ host generation 0
a=candidate:2131708102 2 udp 2113937151 192.168.0.100 56399 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 56400 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 56400 typ host generation 0
a=candidate:1054674330 1 udp 2113937151 192.168.15.174 56401 typ host generation 0
a=candidate:1054674330 2 udp 2113937151 192.168.15.174 56401 typ host generation 0
a=candidate:831304758 1 tcp 1509957375 192.168.0.100 0 typ host generation 0
a=candidate:831304758 2 tcp 1509957375 192.168.0.100 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:1885270378 1 tcp 1509957375 192.168.15.174 0 typ host generation 0
a=candidate:1885270378 2 tcp 1509957375 192.168.15.174 0 typ host generation 0
a=ice-ufrag:T1B5PfyM7DyLrdlr
a=ice-pwd:ZLI+Zz/FhMSEu1p+tAO6yKQo
a=ice-options:google-ice
a=fingerprint:sha-256 2A:D3:03:94:3C:EC:15:3B:FB:0B:91:AF:29:D4:B9:84:64:53:34:3F:DE:5F:0D:0D:87:93:5A:5A:82:67:54:6C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:1Fp438zoCVVcA747GJ7TitPGzGc0Kkwncmi4gLCY
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:nVlpjPhulEmeKPOiK37SrL3Wt3O6DN3mTLqaBnbd
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2109020515 cname:9e1SqHu3kfZlTPr8
a=ssrc:2109020515 msid:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0 FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0a0
a=ssrc:2109020515 mslabel:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0
a=ssrc:2109020515 label:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0a0

jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK2611438;received=192.168.0.100
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
To: sip:1111@192.168.0.101;tag=as6af7e2c4
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5946 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="68ffb0aa"
Content-Length: 0

jssip-0.3.0.js:670
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:1111@192.168.0.101 SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK2611438
To: sip:1111@192.168.0.101;tag=as6af7e2c4
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5946 ACK

jssip-0.3.0.js:519
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:1111@192.168.0.101 SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK899407
Max-Forwards: 69
To: sip:1111@192.168.0.101
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5947 INVITE
Authorization: Digest algorithm=MD5, username=“100”, realm=“asterisk”, nonce=“68ffb0aa”, uri="sip:1111@192.168.0.101", response="fad1e51fbcbf3285c7cc4ca67e57477f"
Contact: sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 2248

v=0
o=- 1211472537477211476 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0
m=audio 56399 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.0.100
a=rtcp:56399 IN IP4 192.168.0.100
a=candidate:2131708102 1 udp 2113937151 192.168.0.100 56399 typ host generation 0
a=candidate:2131708102 2 udp 2113937151 192.168.0.100 56399 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 56400 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 56400 typ host generation 0
a=candidate:1054674330 1 udp 2113937151 192.168.15.174 56401 typ host generation 0
a=candidate:1054674330 2 udp 2113937151 192.168.15.174 56401 typ host generation 0
a=candidate:831304758 1 tcp 1509957375 192.168.0.100 0 typ host generation 0
a=candidate:831304758 2 tcp 1509957375 192.168.0.100 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:1885270378 1 tcp 1509957375 192.168.15.174 0 typ host generation 0
a=candidate:1885270378 2 tcp 1509957375 192.168.15.174 0 typ host generation 0
a=ice-ufrag:T1B5PfyM7DyLrdlr
a=ice-pwd:ZLI+Zz/FhMSEu1p+tAO6yKQo
a=ice-options:google-ice
a=fingerprint:sha-256 2A:D3:03:94:3C:EC:15:3B:FB:0B:91:AF:29:D4:B9:84:64:53:34:3F:DE:5F:0D:0D:87:93:5A:5A:82:67:54:6C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:1Fp438zoCVVcA747GJ7TitPGzGc0Kkwncmi4gLCY
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:nVlpjPhulEmeKPOiK37SrL3Wt3O6DN3mTLqaBnbd
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2109020515 cname:9e1SqHu3kfZlTPr8
a=ssrc:2109020515 msid:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0 FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0a0
a=ssrc:2109020515 mslabel:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0
a=ssrc:2109020515 label:FkfqoJ6N9I0XxdHIuw0SWtNozWnKkw5Wa2C0a0

jssip-0.3.0.js:519
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK2611438 jssip-0.3.0.js:1944
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK899407;received=192.168.0.100
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
To: sip:1111@192.168.0.101
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5947 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1111@192.168.0.101:5060;transport=WS
Content-Length: 0

jssip-0.3.0.js:670
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK899407;received=192.168.0.100
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
To: sip:1111@192.168.0.101;tag=as3981f3f4
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5947 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1111@192.168.0.101:5060;transport=WS
Content-Type: application/sdp
Content-Length: 372

v=0
o=root 420552460 420552460 IN IP4 192.168.0.101
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.0.101
t=0 0
m=audio 13864 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:nA6ICsiDrDVvMhgjIdESO4syo80/DZAJ8z0UjNAt

jssip-0.3.0.js:670
JsSIP | RTC SESSION | stream added: default jssip-0.3.0.js:3360
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:1111@192.168.0.101:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS fhmojhigdgmh.invalid;branch=z9hG4bK988338
Max-Forwards: 69
To: sip:1111@192.168.0.101;tag=as3981f3f4
From: “100” sip:100@192.168.0.101;tag=itiihlqp35
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 5947 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

jssip-0.3.0.js:519
JsSIP | EVENT EMITTER | emitting event started jssip-0.3.0.js:187
JsSIP | TRANSPORT | received WebSocket text message:

BYE sip:ckq0ph3s@fhmojhigdgmh.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.101:5060;branch=z9hG4bK057fecc3
Max-Forwards: 70
From: sip:1111@192.168.0.101;tag=as3981f3f4
To: “100” sip:100@192.168.0.101;tag=itiihlqp35
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.7.0
Proxy-Authorization: Digest username=“r8oumeb5”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.0.101”, nonce=“68ffb0aa”, response="e67da068af1d4c9a8cbf923209b7d7d8"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

jssip-0.3.0.js:670
JsSIP | TRANSPORT | sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.101:5060;branch=z9hG4bK057fecc3
To: “100” sip:100@192.168.0.101;tag=itiihlqp35
From: sip:1111@192.168.0.101;tag=as3981f3f4
Call-ID: pnqo0gtlr1vmeqg8ctin
CSeq: 102 BYE
Content-Length: 0

jssip-0.3.0.js:519
JsSIP | RTC SESSION | closing INVITE session pnqo0gtlr1vmeqg8ctinitiihlqp35 jssip-0.3.0.js:4193
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392
JsSIP | DIALOG | dialog pnqo0gtlr1vmeqg8ctinitiihlqp35as3981f3f4 deleted jssip-0.3.0.js:2543
JsSIP | EVENT EMITTER | emitting event ended jssip-0.3.0.js:187
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK057fecc3 jssip-0.3.0.js:2069
JsSIP | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK899407 jssip-0.3.0.js:1935
JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK899407 jssip-0.3.0.js:1924[/code]

uuid-devel/libuuid-devel was installed.

RTP debug from asterisk:

... Sent RTP packet to 192.168.0.100:56399 (type 08, seq 027930, ts 128800, len 000164) Sent RTP packet to 192.168.0.100:56399 (type 08, seq 027931, ts 128960, len 000164) Sent RTP packet to 192.168.0.100:56399 (type 08, seq 027932, ts 129120, len 000164) Sent RTP packet to 192.168.0.100:56399 (type 08, seq 027933, ts 129280, len 000164) Sent RTP packet to 192.168.0.100:56399 (type 08, seq 027934, ts 129440, len 000164) -- Auto fallthrough, channel 'SIP/100-00000005' status is 'UNKNOWN' vbilling*CLI>

6

Your RTP debug still doesn’t show the ‘via ICE’ string Check if your RTP.conf has icesupport=yes enabled, add an answer as first priority just to make sure the channel answer(I know that playback does that but try it).

7

ICE is required? I turn off ICE in asterisk. Do I have to turn ICE?

8

Yes is required.

9

how to good configure ICE in Asterisk?

10

Add icesupport=yes in your rtp.conf and also per peer in the sip.conf. But if you dont have uuid-devel installed even if you set that it will not work.

11

@avaismo thanks

it work. i installed uuid and uuid-devel :wink: