Cannot create webrtc call using sip.js

Cannot create webrtc call using sip.js - sip

I am not able to create a Webrtc call using sip.js on FreeSWITCH. I believe there are some issues with Freeswitch configuration, but I'm not able to figure out where the issue is and how to figure it out.
Any help would be highly appreciated. I've attached my console log:
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | SessionDescriptionHandlerOptions: {"constraints":{},"peerConnectionOptions":{}}
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | initPeerConnection
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | New peer connection created
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | acquiring local media
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK6799202;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=taqhpgkssv
To: <sip:1002#XYZ.com>;tag=10ga65Ue7mQee
Call-ID: fi260t6dtqnqn0ptskor9r
CSeq: 2907 REGISTER
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
WWW-Authenticate: Digest realm="XYZ.com", nonce="15500d94-ee66-11e8-b1ab-4361709f15e6", algorithm=MD5, qop="auth"
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
REGISTER sip:XYZ.com SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK8071543
Max-Forwards: 70
To: <sip:1002#XYZ.com>
From: <sip:1002#XYZ.com>;tag=taqhpgkssv
Call-ID: fi260t6dtqnqn0ptskor9r
CSeq: 2908 REGISTER
Authorization: Digest algorithm=MD5, username="1002", realm="XYZ.com", nonce="15500d94-ee66-11e8-b1ab-4361709f15e6", uri="sip:XYZ.com", response="5c9ea6e954ab46bce7d4e47c32ca8412", qop=auth, cnonce="3r55378pf6n7", nc=00000001
Contact: <sip:5l366f2t#jttr6lobgmns.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:e6e84559-2720-4d41-ad03-146b793010e2>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:23 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK8071543;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=taqhpgkssv
To: <sip:1002#XYZ.com>;tag=299270cj4XD1S
Call-ID: fi260t6dtqnqn0ptskor9r
CSeq: 2908 REGISTER
Contact: <sip:5l366f2t#jttr6lobgmns.invalid;transport=ws>;expires=600
Date: Thu, 22 Nov 2018 14:51:24 GMT
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | acquired local media streams
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: gathering
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:2174247646 1 udp 2113937151 192.168.3.199 49154 typ host generation 0 ufrag MAkZ network-cost 999
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:2174247646 1 udp 2113937151 192.168.3.199 49156 typ host generation 0 ufrag MAkZ network-cost 999
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:842163049 1 udp 1677729535 146.4.10.10 49156 typ srflx raddr 192.168.3.199 rport 49156 generation 0 ufrag MAkZ network-cost 999
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:25 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:842163049 1 udp 1677729535 146.4.10.10 49154 typ srflx raddr 192.168.3.199 rport 49154 generation 0 ufrag MAkZ network-cost 999
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: complete
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
INVITE sip:1001#XYZ.com SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939
Max-Forwards: 70
To: <sip:1001#XYZ.com>
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:5l366f2t#jttr6lobgmns.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Type: application/sdp
Content-Length: 5285
v=0
o=- 153243296618550758 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf
m=audio 49154 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 146.4.10.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2174247646 1 udp 2113937151 192.168.3.199 49154 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 146.4.10.10 49154 typ srflx raddr 192.168.3.199 rport 49154 generation 0 network-cost 999
a=ice-ufrag:MAkZ
a=ice-pwd:bWXLDz7c+92/dd8QoZStRG2O
a=ice-options:trickle
a=fingerprint:sha-256 D9:03:E8:8E:59:9C:F9:C7:F1:97:1C:2E:8B:9D:AA:8B:22:72:79:0A:DA:20:F5:15:0B:92:01:0D:90:96:2E:BC
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
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:4246339017 cname:S6yAd62x0R59DnAa
a=ssrc:4246339017 msid:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf ff315b12-5a25-40df-bd9c-63d5fa8839dd
a=ssrc:4246339017 mslabel:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf
a=ssrc:4246339017 label:ff315b12-5a25-40df-bd9c-63d5fa8839dd
m=video 49156 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114
c=IN IP4 146.4.10.10
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2174247646 1 udp 2113937151 192.168.3.199 49156 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 146.4.10.10 49156 typ srflx raddr 192.168.3.199 rport 49156 generation 0 network-cost 999
a=ice-ufrag:MAkZ
a=ice-pwd:bWXLDz7c+92/dd8QoZStRG2O
a=ice-options:trickle
a=fingerprint:sha-256 D9:03:E8:8E:59:9C:F9:C7:F1:97:1C:2E:8B:9D:AA:8B:22:72:79:0A:DA:20:F5:15:0B:92:01:0D:90:96:2E:BC
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 x-google-profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 H264/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=124
a=rtpmap:123 red/90000
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=123
a=rtpmap:114 ulpfec/90000
a=ssrc-group:FID 2475933144 456734105
a=ssrc:2475933144 cname:S6yAd62x0R59DnAa
a=ssrc:2475933144 msid:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf 2c31a0cb-b321-42e0-891b-d077b29416ce
a=ssrc:2475933144 mslabel:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf
a=ssrc:2475933144 label:2c31a0cb-b321-42e0-891b-d077b29416ce
a=ssrc:456734105 cname:S6yAd62x0R59DnAa
a=ssrc:456734105 msid:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf 2c31a0cb-b321-42e0-891b-d077b29416ce
a=ssrc:456734105 mslabel:KhwBRkNdQhKG8uBiFShsdLtQDpQghHNP4DCf
a=ssrc:456734105 label:2c31a0cb-b321-42e0-891b-d077b29416ce
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate gathering complete
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 100 Trying
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
User-Agent: UCP SIP
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:1001#21.135.71.140:5070;transport=udp>
User-Agent: UCP SIP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "Outbound Call" <sip:dv7k2aea#XYZ.com>;party=calling;privacy=off;screen=no
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:26 GMT+0100 (Central European Standard Time) | sip.dialog | new UAC dialog created with status EARLY
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:31 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:1001#21.135.71.140:5070;transport=udp>
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1611
Remote-Party-ID: "Outbound Call" <sip:dv7k2aea#XYZ.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1542875692 1542875693 IN IP4 21.135.71.140
s=FreeSWITCH
c=IN IP4 21.135.71.140
t=0 0
a=msid-semantic: WMS PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
m=audio 22600 UDP/TLS/RTP/SAVPF 111 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:106 CN/8000
a=ptime:20
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:22600 IN IP4 21.135.71.140
a=ice-ufrag:sLb4c6Ntf3hGXrOU
a=ice-pwd:fWOE8ym4ReEccY5uz6pC33V8
a=candidate:7384996702 1 udp 659136 21.135.71.140 22600 typ host generation 0
a=end-of-candidates
a=ssrc:2482680622 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:2482680622 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh a0
a=ssrc:2482680622 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:2482680622 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kha0
m=video 21928 UDP/TLS/RTP/SAVPF 98
b=AS:1024
a=rtpmap:98 VP9/90000
a=fmtp:98 x-google-profile-id=0
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:21928 IN IP4 21.135.71.140
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=ssrc:1711252375 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:1711252375 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh v0
a=ssrc:1711252375 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:1711252375 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3khv0
a=ice-ufrag:yysGP93QpcAYPz41
a=ice-pwd:pNHITqOLemWhNzScXbJCaCjq
a=candidate:0811010037 1 udp 659136 21.135.71.140 21928 typ host generation 0
a=end-of-candidates
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:31 GMT+0100 (Central European Standard Time) | sip.dialog | dialog d1gsfqrmq28p029ntjsg9ktrgj6ju04UvmBQeSyFt6g changed to CONFIRMED state
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:31 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | track added
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:36 GMT+0100 (Central European Standard Time) | sip.invitecontext.sessionDescriptionHandler | track added
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
ACK sip:1001#21.135.71.140:5070;transport=udp SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK5822553
Max-Forwards: 70
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 ACK
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:1001#21.135.71.140:5070;transport=udp>
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1611
Remote-Party-ID: "Outbound Call" <sip:dv7k2aea#XYZ.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1542875692 1542875693 IN IP4 21.135.71.140
s=FreeSWITCH
c=IN IP4 21.135.71.140
t=0 0
a=msid-semantic: WMS PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
m=audio 22600 UDP/TLS/RTP/SAVPF 111 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:106 CN/8000
a=ptime:20
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:22600 IN IP4 21.135.71.140
a=ice-ufrag:sLb4c6Ntf3hGXrOU
a=ice-pwd:fWOE8ym4ReEccY5uz6pC33V8
a=candidate:7384996702 1 udp 659136 21.135.71.140 22600 typ host generation 0
a=end-of-candidates
a=ssrc:2482680622 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:2482680622 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh a0
a=ssrc:2482680622 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:2482680622 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kha0
m=video 21928 UDP/TLS/RTP/SAVPF 98
b=AS:1024
a=rtpmap:98 VP9/90000
a=fmtp:98 x-google-profile-id=0
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:21928 IN IP4 21.135.71.140
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=ssrc:1711252375 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:1711252375 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh v0
a=ssrc:1711252375 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:1711252375 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3khv0
a=ice-ufrag:yysGP93QpcAYPz41
a=ice-pwd:pNHITqOLemWhNzScXbJCaCjq
a=candidate:0811010037 1 udp 659136 21.135.71.140 21928 typ host generation 0
a=end-of-candidates
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
ACK sip:1001#21.135.71.140:5070;transport=udp SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK5822553
Max-Forwards: 70
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 ACK
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:1001#21.135.71.140:5070;transport=udp>
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1611
Remote-Party-ID: "Outbound Call" <sip:dv7k2aea#XYZ.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1542875692 1542875693 IN IP4 21.135.71.140
s=FreeSWITCH
c=IN IP4 21.135.71.140
t=0 0
a=msid-semantic: WMS PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
m=audio 22600 UDP/TLS/RTP/SAVPF 111 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:106 CN/8000
a=ptime:20
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:22600 IN IP4 21.135.71.140
a=ice-ufrag:sLb4c6Ntf3hGXrOU
a=ice-pwd:fWOE8ym4ReEccY5uz6pC33V8
a=candidate:7384996702 1 udp 659136 21.135.71.140 22600 typ host generation 0
a=end-of-candidates
a=ssrc:2482680622 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:2482680622 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh a0
a=ssrc:2482680622 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:2482680622 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kha0
m=video 21928 UDP/TLS/RTP/SAVPF 98
b=AS:1024
a=rtpmap:98 VP9/90000
a=fmtp:98 x-google-profile-id=0
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:21928 IN IP4 21.135.71.140
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=ssrc:1711252375 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:1711252375 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh v0
a=ssrc:1711252375 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:1711252375 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3khv0
a=ice-ufrag:yysGP93QpcAYPz41
a=ice-pwd:pNHITqOLemWhNzScXbJCaCjq
a=candidate:0811010037 1 udp 659136 21.135.71.140 21928 typ host generation 0
a=end-of-candidates
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
ACK sip:1001#21.135.71.140:5070;transport=udp SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK5822553
Max-Forwards: 70
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 ACK
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK9014939;received=146.4.10.10;rport=53279
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 INVITE
Contact: <sip:1001#21.135.71.140:5070;transport=udp>
User-Agent: UCP SIP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1611
Remote-Party-ID: "Outbound Call" <sip:dv7k2aea#XYZ.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1542875692 1542875693 IN IP4 21.135.71.140
s=FreeSWITCH
c=IN IP4 21.135.71.140
t=0 0
a=msid-semantic: WMS PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
m=audio 22600 UDP/TLS/RTP/SAVPF 111 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:106 CN/8000
a=ptime:20
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:22600 IN IP4 21.135.71.140
a=ice-ufrag:sLb4c6Ntf3hGXrOU
a=ice-pwd:fWOE8ym4ReEccY5uz6pC33V8
a=candidate:7384996702 1 udp 659136 21.135.71.140 22600 typ host generation 0
a=end-of-candidates
a=ssrc:2482680622 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:2482680622 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh a0
a=ssrc:2482680622 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:2482680622 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kha0
m=video 21928 UDP/TLS/RTP/SAVPF 98
b=AS:1024
a=rtpmap:98 VP9/90000
a=fmtp:98 x-google-profile-id=0
a=fingerprint:sha-256 3D:31:63:0A:35:84:28:91:9E:94:0F:6A:5A:E2:99:1A:DC:AA:36:4C:86:81:C7:EE:72:EB:0D:F4:A8:87:2E:AA
a=setup:active
a=rtcp-mux
a=rtcp:21928 IN IP4 21.135.71.140
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=ssrc:1711252375 cname:6Xi4zlTR5s2Kv7t2
a=ssrc:1711252375 msid:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh v0
a=ssrc:1711252375 mslabel:PcNb7LcZsUJkOdfTGusiMA5SriqXB3kh
a=ssrc:1711252375 label:PcNb7LcZsUJkOdfTGusiMA5SriqXB3khv0
a=ice-ufrag:yysGP93QpcAYPz41
a=ice-pwd:pNHITqOLemWhNzScXbJCaCjq
a=candidate:0811010037 1 udp 659136 21.135.71.140 21928 typ host generation 0
a=end-of-candidates
sip-0.11.6.js:516 Thu Nov 22 2018 15:51:37 GMT+0100 (Central European Standard Time) | sip.transport | sending WebSocket message:
ACK sip:1001#21.135.71.140:5070;transport=udp SIP/2.0
Via: SIP/2.0/WSS jttr6lobgmns.invalid;branch=z9hG4bK5822553
Max-Forwards: 70
To: <sip:1001#XYZ.com>;tag=4UvmBQeSyFt6g
From: <sip:1002#XYZ.com>;tag=9ktrgj6ju0
Call-ID: d1gsfqrmq28p029ntjsg
CSeq: 1865 ACK
Supported: outbound
User-Agent: SIP.js/0.11.6
Content-Length: 0

Related

Why video call terminated after answer in Flutter WebRTC?

I do video call from https://tryit.jssip.net/ to Flutter, this browser console invite:
INVITE sip:046541#example.com SIP/2.0
Via: SIP/2.0/WSS pvngo3rdjrg0.invalid;branch=z9hG4bK1238648
Max-Forwards: 69
To: <sip:046541#example.com>
From: "morfair-work-pc" <sip:691239#example.com>;tag=gba3bp7ub6
Call-ID: hqsmog5cf3hi7fhiad1b
CSeq: 9239 INVITE
Contact: <sip:flnilmjh#pvngo3rdjrg0.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.8.2
Content-Length: 7287
v=0
o=- 3916041425617185783 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm
m=audio 49474 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 1.1.1.1
a=rtcp:63644 IN IP4 1.1.1.1
a=candidate:4087335268 1 udp 2122260223 10.10.6.5 54981 typ host generation 0 network-id 1
a=candidate:4087335268 2 udp 2122260222 10.10.6.5 54982 typ host generation 0 network-id 1
a=candidate:2970761196 2 udp 1686052606 1.1.1.1 63644 typ srflx raddr 10.10.6.5 rport 54982 generation 0 network-id 1
a=candidate:2970761196 1 udp 1686052607 1.1.1.1 49474 typ srflx raddr 10.10.6.5 rport 54981 generation 0 network-id 1
a=candidate:3172742548 1 tcp 1518280447 10.10.6.5 9 typ host tcptype active generation 0 network-id 1
a=candidate:3172742548 2 tcp 1518280446 10.10.6.5 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:3Qwu
a=ice-pwd:oEwEfLlKoLaN0NNofZrO3Xl7
a=ice-options:trickle
a=fingerprint:sha-256 D0:DD:2B:F4:CB:39:95:43:AA:DB:51:AD:67:D8:6D:31:94:91:4D:71:03:F1:16:6F:F7:A9:3B:70:CE:1C:83:3F
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:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm 110093ea-09f9-4a6a-8191-93256b94a1d6
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:1662500094 cname:brFpXZ/mqc967pVG
a=ssrc:1662500094 msid:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm 110093ea-09f9-4a6a-8191-93256b94a1d6
a=ssrc:1662500094 mslabel:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm
a=ssrc:1662500094 label:110093ea-09f9-4a6a-8191-93256b94a1d6
m=video 16655 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116
c=IN IP4 1.1.1.1
a=rtcp:37889 IN IP4 1.1.1.1
a=candidate:4087335268 1 udp 2122260223 10.10.6.5 54983 typ host generation 0 network-id 1
a=candidate:4087335268 2 udp 2122260222 10.10.6.5 54984 typ host generation 0 network-id 1
a=candidate:2970761196 2 udp 1686052606 1.1.1.1 37889 typ srflx raddr 10.10.6.5 rport 54984 generation 0 network-id 1
a=candidate:2970761196 1 udp 1686052607 1.1.1.1 16655 typ srflx raddr 10.10.6.5 rport 54983 generation 0 network-id 1
a=candidate:3172742548 1 tcp 1518280447 10.10.6.5 9 typ host tcptype active generation 0 network-id 1
a=candidate:3172742548 2 tcp 1518280446 10.10.6.5 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:3Qwu
a=ice-pwd:oEwEfLlKoLaN0NNofZrO3Xl7
a=ice-options:trickle
a=fingerprint:sha-256 D0:DD:2B:F4:CB:39:95:43:AA:DB:51:AD:67:D8:6D:31:94:91:4D:71:03:F1:16:6F:F7:A9:3B:70:CE:1C:83:3F
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
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:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm dbf0c622-ff08-437d-b291-5d2c39ae84d9
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:35 AV1X/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f
a=rtpmap:118 rtx/90000
a=fmtp:118 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 1817057936 1383878196
a=ssrc:1817057936 cname:brFpXZ/mqc967pVG
a=ssrc:1817057936 msid:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm dbf0c622-ff08-437d-b291-5d2c39ae84d9
a=ssrc:1817057936 mslabel:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm
a=ssrc:1817057936 label:dbf0c622-ff08-437d-b291-5d2c39ae84d9
a=ssrc:1383878196 cname:brFpXZ/mqc967pVG
a=ssrc:1383878196 msid:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm dbf0c622-ff08-437d-b291-5d2c39ae84d9
a=ssrc:1383878196 mslabel:TaUfv8YlphfrxofRLtlUsNbzdm02ObsHDcZm
a=ssrc:1383878196 label:dbf0c622-ff08-437d-b291-5d2c39ae84d9
My app from Asterisk get this INVITE body:
v=0
o=- 1291471226 1291471226 IN IP4 45.139.24.57
s=Asterisk
c=IN IP4 45.139.24.57
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0 video-1
m=audio 13462 UDP/TLS/RTP/SAVPF 8 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 60:C8:FC:AA:01:2A:CE:0B:BE:C3:94:D9:0C:49:92:75:06:56:2D:F8:ED:93:B0:D4:27:C2:E4:88:6C:2F:13:58
a=ice-ufrag:7d6973e77e596a7c5ca975fd1ee04a4f
a=ice-pwd:78d20512467e79ab088da2a9631ae4db
a=candidate:H2d8b1839 1 UDP 2130706431 45.139.24.57 13462 typ host
a=candidate:H89e136a7 1 UDP 2130706431 2a0f:1140:100::1c 13462 typ host
a=candidate:He0ecc6ac 1 UDP 2130706431 fe80::f816:3eff:fe49:f4a8 13462 typ host
a=candidate:H63d27d08 1 UDP 2130706431 fe80::42:4fff:feb8:99ab 13462 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:302749605 cname:df664cb4-f1ec-4cfa-92e0-8fd62d07df8a
a=msid:64715997-bdfb-4395-ac9e-974583f77ede 6f0d0502-9a0b-432c-a302-7de6ad4ce5f1
a=rtcp-fb:* transport-cc
a=mid:audio-0
m=video 13462 UDP/TLS/RTP/SAVPF 99
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 60:C8:FC:AA:01:2A:CE:0B:BE:C3:94:D9:0C:49:92:75:06:56:2D:F8:ED:93:B0:D4:27:C2:E4:88:6C:2F:13:58
a=ice-ufrag:7d6973e77e596a7c5ca975fd1ee04a4f
a=ice-pwd:78d20512467e79ab088da2a9631ae4db
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=sendrecv
a=rtcp-mux
a=ssrc:608170838 cname:de6c7075-8495-4f92-a5b5-a82c999b284e
a=msid:3ab0f427-0931-48f5-9504-2cbd21395e0d 205bee32-2106-458b-b238-4eb32f73e1ea
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:video-1
After answer on mobile app call terminated with answer from mobile app:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS pvngo3rdjrg0.invalid;rport=56438;received=127.0.0.1;branch=z9hG4bK7851071
Call-ID: hqsmol3kkjind0o2qiqe
From: "morfair-work-pc" <sip:691239#example.com>;tag=imv2vthavd
To: <sip:046541#example.com>;tag=785029d9-6f99-4ebb-9971-d4856dbd9494
CSeq: 5873 INVITE
Server: Asterisk PBX 18.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=58
Content-Length: 0
Browser show error: Incompatible SDP.
In debug console I see errors:
I/flutter (28279): [2021-11-18 12:18:05.217] Level.debug rtc_session.dart:650 ::: emit "sdp"
W/VideoCapabilities(28279): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities(28279): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities(28279): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities(28279): Unrecognized profile 2130706434 for video/avc
...
I/flutter (28279): [2021-11-18 12:18:05.265] Level.debug rtc_session.dart:2941 ::: session failed
I/flutter (28279): [2021-11-18 12:18:05.268] Level.debug rtc_session.dart:2944 ::: emit "_failed"
I/flutter (28279): [2021-11-18 12:18:05.272] Level.debug rtc_session.dart:1474 ::: close()
I/flutter (28279): [2021-11-18 12:18:05.276] Level.debug rtc_session.dart:2955 ::: emit "failed"
I/flutter (28279): [2021-11-18 12:18:05.279] Level.debug sip_ua_helper.dart:214 ::: call failed with cause: Code: [488], Cause: WebRTC Error, Reason: SetRemoteDescription(offer) failed
I/flutter (28279): MySipHelperListener callStateChanged() state: CallStateEnum.FAILED
I/flutter (28279): [2021-11-18 12:18:05.284] Level.error rtc_session.dart:660 ::: emit "peerconnection:setremotedescriptionfailed" [error:Unable to RTCPeerConnection::setRemoteDescription: peerConnectionSetRemoteDescription(): WEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='video-1'.]
E/flutter (28279): [ERROR:flutter/lib/ui/ui_dart_state.cc(209)] Unhandled Exception: Assertion failed: "peerconnection.setRemoteDescription() failed"
E/flutter (28279): #0 RTCSession.answer (package:sip_ua/src/rtc_session.dart:663:7)
E/flutter (28279): <asynchronous suspension>
E/flutter (28279):
I/AudioManager(28279): setMode mode:0
I/AudioManager(28279): abandonAudioFocus
I/flutter (28279): [2021-11-18 12:18:05.311] Level.debug rtc_session.dart:1492 ::: close() | closing local MediaStream
I have two mobile devices: Redmi 9A and Huawei Nova LTE (CAN-L11). And two browsers: Firefox 94.0.2 and Chrome 96.0.4664.45. Site: https://tryit.jssip.net/.
And only call from Firefox to Huawei CAN-L11 work!
Chrome -> Redmi 9A - Incompatible SDP
Firefox -> Redmi 9A - Incompatible SDP
Chrome -> Huawei CAN-L11 - Incompatible SDP
Firefox -> Huawei CAN-L11 - WORK!!!!
P.S.: Call from mobile to browser always works.

SIP Route parameters: r2 and rpp

I'm using the VoIP provider Messagenet and I see that it sends this INVITE to my client:
INVITE sip:me#my.ip.address;transport=udp SIP/2.0
Record-Route: <sip:212.97.59.76:5061;r2=on;lr=on;ftag=as2a6c9e96;rpp=np>
Record-Route: <sip:212.97.59.76;r2=on;lr=on;ftag=as2a6c9e96;rpp=np>
Via: SIP/2.0/UDP 212.97.59.76:5061;branch=z9hG4bKb812.a8e25f17.0
Via: SIP/2.0/UDP 193.227.104.21:5060;branch=z9hG4bK1966d0a8
Max-Forwards: 69
From: "fromname" <sip:fromnumber#sip.messagenet.it>;tag=as2a6c9e96
To: <sip:mynumber#212.97.59.76>
Contact: <sip:fromnumber#193.227.104.21:5060>
Call-ID: 46cee45e062a30e3372663265a0be595#sip.messagenet.it
CSeq: 102 INVITE
User-Agent: whisky
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Mnet-InLeg: whisky:1530689207.16483286;NH9HCiZ0TqZGK1ACkkhlvh67g1ig7LPEf+W5lblauyjqYqxgm1B2stIr6Mog/CC6
Content-Type: application/sdp
Content-Length: 377
v=0
o=root 60186075 60186075 IN IP4 193.227.104.23
s=whisky
c=IN IP4 193.227.104.23
t=0 0
m=audio 50216 RTP/AVP 18 8 3 0 97 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
My question is: what do the parameters r2 and rpp mean in the Record-Route headers?
Record-Route: <sip:212.97.59.76:5061;r2=on;lr=on;ftag=as2a6c9e96;rpp=np>
Record-Route: <sip:212.97.59.76;r2=on;lr=on;ftag=as2a6c9e96;rpp=np>
I'm looking for some documentations about this, but I have found nothing until now.

Liblinphone SIP Error: 403 Phone Check Failed

I use the open source sip library liblinphone to make a outgoing call in Swift:
func inviteCall(lc: COpaquePointer){
let identity = "sip:07XXXXXX#XXX.XXX.XXX.XX:5060"
let callee = linphone_address_new(identity)
var call: COpaquePointer = linphone_core_invite_address(lc, callee)
for _ in 1...20{
linphone_core_iterate(lc); /* first iterate initiates registration */
ms_usleep(1000 * 1000);
}
}
I got the following eror meesage from SIP Server:
2016-06-09 10:38:01:490 ortp-message-channel [0x7ffe73845000]: received [315] new bytes from [UDP://1.1.1.1:5060]:
SIP/2.0 403 Phone Check Failed
Via: SIP/2.0/UDP 172.20.10.2:5060;branch=z9hG4bK.wT4sI9IJA;rport=61355;received=180.217.232.152
From: <sip:XXXXXX#1.1.1.1>;tag=q6lu03sDI
To: sip:XXXXXX#1.1.1.1;tag=00ddf5d9798df559a35d085b6da2ca8e.8b81
CSeq: 21 INVITE
Call-ID: al9nF3pfWh
Content-Length: 0
I googled about 403 Phone Check Failed, but not mucn information available. Any idea how to troubleshoot it ? Thanks.
PS: Account/IP information are masked
This was the invite message sent to SIP Server
INVITE sip:XXXXXXX216#XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.2:5060;branch=z9hG4bK.wT4sI9IJA;rport
From: <sip:XXXXXXX2519#XXX.XXX.XXX.XXX>;tag=q6lu03sDI
To: sip:XXXXXXX216#XXX.XXX.XXX.XXX
CSeq: 21 INVITE
Call-ID: al9nF3pfWh
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 473
Contact: <sip:XXXXXXX2519#180.217.232.152:61355>;+sip.instance="<urn:uuid:5b5a5dd0-5447-4066-9a3a-b652eb4db075>"
User-Agent: (belle-sip/1.4.2)
Proxy-Authorization: Digest realm="XXX.XXX.XXX.XXX", nonce="5758d7b1e9287b8975df56f3b8a5e9f3f455b585", username="XXXXXXX2519", uri="sip:XXXXXXX216#XXX.XXX.XXX.XXX:5060", response="cf978e8e12c032aafa9a03e5d9450e9f", cnonce="7fb0fe3c", nc=00000001, qop=auth
v=0
o=XXXXXXX2519 238 1151 IN IP4 172.20.10.2
s=Talk
c=IN IP4 172.20.10.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 5000

I found the problem. We can't do register and invite sip command in the same time. I think this caused the issue.

Asterisk failed to deliver sound on LTE(4G) network

I've installed Asterisk 11, and two wifi phones are fine to talk through asterisk server. However, a wifi phone and LTE(4G) phone can't deliver sounds.
Asterisk sip.conf
[general]
context=default ; Default context for incoming calls
bindport=5060 ; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw
register => 12121111111:1234:11111111#sipauth.deltathree.com/1000
srvlookup=no
directrtpsetup=yes
trustpid=yes
sendrpid=no
qualify=yes
callevents=yes
insecure=invite
pedantic=no
videosupport=yes
canreinvite=yes
nat=yes
externip=XXX.XXX.91.12
localnet=10.7.21.4/255.255.255.0
qualify=yes
directmedia=yes
Sip settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: Yes
User Agent: Asterisk PBX 11.8.1
SDP Session Name: Asterisk PBX 11.8.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Enabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Enabled using externhost
Externhost: XXX.52.91.12:0
Externaddr: XXX.52.91.12:0
Externrefresh: 600
Localnet: XX.7.21.0/255.255.255.0
XX.7.21.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 2000
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
Realtime SIP Settings:
----------------------
Realtime Peers: Yes
Realtime Regs: No
Cache Friends: No
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120 (Disabled)
sip logs
When I look at sip logs, it looks like fine. I just see one more "invite" from server to wifi-phone.
interface: eth0 (10.7.21.0/255.255.255.0)
filter: ( port 5060 ) and (ip or ip6)
#
U 2014/04/16 22:46:28.514023 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
INVITE sip:2000#asterisk-sip-domain.com SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;rport.
From: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000#asterisk-sip-domain.com>.
CSeq: 20 INVITE.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 372.
Contact: <sip:1000#//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.
v=0.
o=1000 2350 2859 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 124 120 111 110 0 8 101.
a=rtpmap:124 opus/48000.
a=fmtp:124 useinbandfec=1; usedtx=1.
a=rtpmap:120 SILK/16000.
a=rtpmap:111 speex/16000.
a=fmtp:111 vbr=on.
a=rtpmap:110 speex/8000.
a=fmtp:110 vbr=on.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:28.517399 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000#asterisk-sip-domain.com>.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.
#
U 2014/04/16 22:46:28.522887 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000#//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 2014/04/16 22:46:29.022450 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000#//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 2014/04/16 22:46:29.113047 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: sip:2000#//LTE-PHONE-PUBLIC-IP//:63968.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
.
#
U 2014/04/16 22:46:29.426139 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
#
U 2014/04/16 22:46:29.426158 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
#
U 2014/04/16 22:46:29.427976 f:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.
** (WHY IT MAKES ONE MORE INVITE FROM SERVER TO WIFI-PHONE???)**
#
U 2014/04/16 22:46:30.307448 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:48504
INVITE sip:1000#//WIFI-PUBLIC-IP//:48504 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK451726cf;rport.
Max-Forwards: 70.
From: <sip:2000#//AMAZON-EC2-SERVER//>;tag=as30b8a8a5.
To: <sip:1000#//WIFI-PUBLIC-IP//:48504>.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: 1c2fd2cd6a4ac372408845e8077ba2b5#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 741350827 741350827 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 2014/04/16 22:46:30.816230 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>;+sip.instance="<urn:uuid:8afceca3-368f-4f57-a586-6056d3492371>".
Content-Type: application/sdp.
Content-Length: 183.
.
v=0.
o=2000 2310 1562 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Talk.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
b=AS:380.
t=0 0.
m=audio 30390 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:30.816888 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
ACK sip:2000#//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK680dd0d2;rport.
Max-Forwards: 70.
From: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000#//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Contact: <sip:1000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.
#
U 2014/04/16 22:46:30.817278 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 1551912347 1551912347 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 2014/04/16 22:46:30.925455 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
ACK sip:2000#//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;rport;branch=z9hG4bK.qV8rz6rI4.
From: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
CSeq: 20 ACK.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
.
#
U 2014/04/16 22:46:35.277987 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
BYE sip:1000#//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;rport.
From: <sip:2000#//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
CSeq: 111 BYE.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
Max-Forwards: 70.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.
#
U 2014/04/16 22:46:35.278525 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;received=//LTE-PHONE-PUBLIC-IP//;rport=63968.
From: <sip:2000#//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000#//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
Call-ID: 60d3866362c2076357b37d2d4b930652#//AMAZON-EC2-SERVER//:5060.
CSeq: 111 BYE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.278797 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
INVITE sip:1000#//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
Max-Forwards: 70.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1551912347 1551912348 IN IP4 //AMAZON-EC2-SERVER//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //AMAZON-EC2-SERVER//.
t=0 0.
m=audio 19500 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 2014/04/16 22:46:35.418765 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
.
#
U 2014/04/16 22:46:35.441248 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:1000#//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
Content-Type: application/sdp.
Content-Length: 180.
.
v=0.
o=1000 2350 2861 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:35.441661 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
ACK sip:1000#//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK7dd12d7d;rport.
Max-Forwards: 70.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000#//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.441754 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
BYE sip:1000#//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
Max-Forwards: 70.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.474403 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
From: "........." <sip:2000#asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000#asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
exit
21 received, 0 dropped
Do you see why it failed to deliver sound when a device is on LTE(4G) network?

I changed below code in sip.conf and user's conf.
canreinvite = yes
It all worked fine. However, it delivers sound through Asterisk Server, which means the server has to take care of voice traffic.

SIP Invite content-length

I'm developing a SIP provider application.I use transport UDP.And I have a Questions,
I sending a Invite message SIP Server.
INVITE sip:102#192.168.1.33 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5001;branch=z9hG4bK9232c352-a28c-4467-988e-8027e0031209;rport
To: <sip:102#192.168.1.33>
From: "101"<sip:101#192.168.1.33:5060>;tag=rkktjbvq
CSeq: 1 INVITE
Call-ID: lrfnpvlvrbojabxnuldgejvncshccjpwsfxsobpcpmjrnsvkeh
Max-Forwards: 70
Contact: <sip:101#192.168.1.33:5001>
User-Agent: Iconium
Content-Type: application/sdp
Content-Length: 849
v=0
o=101 940412967 940412967 IN IP4 192.168.1.33
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.1.33
t=0 0
m=audio 5003 RTP/AVP 8 0 101 98 9 3 100 97 103 15 4 104 105 106 107 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64000
a=rtpmap:3 GSM/8000
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:103 L16/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=rtpmap:104 G726-16/8000
a=rtpmap:105 G726-24/8000
a=rtpmap:106 G726-32/8000
a=rtpmap:107 G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 5005 RTP/AVP 102 99 34
a=rtpmap:102 H263-1998/90000
a=fmtp:102 QCIF=1;CIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1
a=sendrecv
Everything is okay.And SIP Server response with Proxy-Authenticate
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.33:5001;branch=z9hG4bK9232c352-a28c-4467-988e-8027e0031209;rport=5001
Proxy-Authenticate:Digest nonce="414d535c05ab5fd821:79225947c170510b155be0828d92e7e4", algorithm=MD5, realm="3CXPhoneSystem"
To: <sip:102#192.168.1.33>;tag=d92fe85a
From: "101"<sip:101#192.168.1.33:5060>;tag=rkktjbvq
Call-ID: lrfnpvlvrbojabxnuldgejvncshccjpwsfxsobpcpmjrnsvkeh
CSeq: 1 INVITE
User-Agent: 3CXPhoneSystem 10.0.23053.0
Content-Length: 0
And I send Authanticate with ACK;
INVITE sip:102#192.168.1.33 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5001;branch=z9hG4bKaaf896d5-bd08-48f2-8e3d-0cf043e8324e;rport
To: <sip:102#192.168.1.33>
From: "101"<sip:101#192.168.1.33:5060>;tag=rkktjbvq
CSeq: 2 INVITE
Call-ID: lrfnpvlvrbojabxnuldgejvncshccjpwsfxsobpcpmjrnsvkeh
Max-Forwards: 70
Contact: <sip:101#192.168.1.33:5001>
User-Agent: Iconium
Content-Type: application/sdp
Proxy-Authorization:Digest username="101", realm="3CXPhoneSystem", nonce="414d535c05ab5fd821:79225947c170510b155be0828d92e7e4", response="8592afb1b7f3440afd9607dc3db588cb", uri="sip:102#192.168.1.33", algorithm=MD5
Content-Length: 849
v=0
o=101 940412967 940412967 IN IP4 192.168.1.33
s=Ozeki VoIP SIP SDK
c=IN IP4 192.168.1.33
t=0 0
m=audio 5003 RTP/AVP 8 0 101 98 9 3 100 97 103 15 4 104 105 106 107 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64000
a=rtpmap:3 GSM/8000
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:103 L16/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=rtpmap:104 G726-16/8000
a=rtpmap:105 G726-24/8000
a=rtpmap:106 G726-32/8000
a=rtpmap:107 G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 5005 RTP/AVP 102 99 34
a=rtpmap:102 H263-1998/90000
a=fmtp:102 QCIF=1;CIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1
a=sendrecv
And Sıp server response with
SIP 2.0/400 Bad Content-Length(larger than datagram)
But in Ozeki Voip SDK ,
while content-length=851 everything is okay.(using UDP transport)
How can I do for this?Where is the my problem?
Edit:
My UDP client:
private void SendSIPMessage(string mesaj, bool korumalı)
{
IPEndPoint remotendpoint = new IPEndPoint(IPAddress.Any, port);
UdpClient udpClient = new UdpClient();
try
{
udpClient.Connect("192.168.1.33", 5060);
Byte[] sendBytes = Encoding.ASCII.GetBytes(mesaj);
udpClient.Send(sendBytes, sendBytes.Length);
string receivedMessage = string.Empty;
Byte[] receiveBytes = udpClient.Receive(ref remotendpoint);
receivedMessage = Encoding.ASCII.GetString(receiveBytes);
label1.Text += receivedMessage + "\n";
}

It might be because your UDP SIP packet is over MTU or 1300 bytes.
RFC 3261 - 18.1.1 Sending Requests
If a request is within 200 bytes of the path MTU, or if it is larger
than 1300 bytes and the path MTU is unknown, the request MUST be sent
using an RFC 2914 [43] congestion controlled transport protocol, such
as TCP.

Simply the server might have a setting to reject too large UDP packets.
I would suggest to remove some codecs from your software. Your list is too large and I don't think that it has any reason.

Most SIP proxies would accept that size of message over UDP no problem. You might want to try a different SIP server. In production systems, it is fairly normal for the SIP message to get larger than a single MTU but this still work when the UDP packets gets fragmented into two parts.

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