WebRTC Sound Problems

Asterisk Asterisk Support
1

Hi, my name is Gerald and I am try to use JSSIP and WebRTC, but we do not receive audio from calls.

My instalation is CentOS 6.5 (Linux mercurio 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux) with Asterisk 11.9.0 built by root @ mercurio on a i686 running Linux on 2014-04-23 22:24:19 UTC.

I configure all sip.conf/rtp.conf/htp.conf, I can register but when I try to make a call it show me at the SIP debug

<— SIP read from WS:189.232.221.54:63589 —>
BYE sip:1000@187.122.82.197:0;transport=ws SIP/2.0
Via: SIP/2.0/WS mmj0ua0tjbct.invalid;branch=z9hG4bK1767708
Max-Forwards: 69
To: sip:1000@189.232.221.54;tag=as1514d3d2
From: “G” sip:8001@189.232.221.54;tag=c0ujquk1fj
Call-ID: 6n90l9o94cd2b5bcltbk
CSeq: 7553 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

I would like some help to solve this issue!

2

Not acceptable here using webrtc usually is caused by a misconfiguration in the sip peer. You need to provide the real configuration for SIP, JSSIP and the debugs from asterisk(sip debug) and web browser(JavaScript debug).

3

Thank you for the help, here are the files.

SIP.CONF

[code]; SIP Configuration example for Asterisk

[general]
context=public ; Default context for incoming calls. Defaults to 'default’
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
realm=webrtc.no-ip.org ; Realm for digest authentication

udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

transport=udp,ws ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.

localnet=192.168.192.7/255.255.255.0
externaddr=webrtc.no-ip.org
nat=auto_force_rport,auto_comedia
directmedia=yes ; Asterisk by default tries to redirect the

[1234]
type=friend
regexten=1234 ; When they register, create extension 1234
secret=______
callerid=“AAAAAA” <1234>
nat=force_rport,comedia
host=dynamic ; This device needs to register
directmedia=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
registertrying=yes ; Send a 10

[8001]
context=default
trustrpid=yes
sendrpid=no
qualify=yes
qualifyfreq=600
type=friend ; we only want to call out, not be called
regexten=8001
secret=_____
nat=auto_force_rport,auto_comedia
encryption=yes
remotesecret=_____
defaultuser=8001 ; Authentication user for outbound proxies
fromuser=8001 ; Many SIP providers require this!
host=dynamic
avpf=yes
icesupport=yes
directmedia=no
dial=SIP/8001
disallow=all
allow=ulaw
[/code]

SIP.LOG

[code]<— SIP read from WS:189.232.221.54:56882 —>
REGISTER sip:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
Call-ID: 1jtu7lhr09rhmilv2k6sgu
CSeq: 81 REGISTER
Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

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

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9
Call-ID: 1jtu7lhr09rhmilv2k6sgu
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=“webrtc.no-ip.org”, nonce="5a4ecdb4"
Content-Length: 0

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

<— SIP read from WS:189.232.221.54:56882 —>
REGISTER sip:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
Call-ID: 1jtu7lhr09rhmilv2k6sgu
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“5a4ecdb4”, uri=“sip:webrtc.no-ip.org”, response="8df3a2d05686d9bc7bc2b856bbe5fba0"
Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

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

OPTIONS sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34
Max-Forwards: 70
From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96
To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws
Contact: sip:8001@192.168.192.7:5060;transport=WS
Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Thu, 24 Apr 2014 20:58:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9
Call-ID: 1jtu7lhr09rhmilv2k6sgu
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:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;expires=600
Date: Thu, 24 Apr 2014 20:58:57 GMT
Content-Length: 0

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

[Khaxixe*CLI>
[0KScheduling destruction of SIP dialog ‘1jtu7lhr09rhmilv2k6sgu’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:189.232.221.54:56882 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34
To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;tag=p9g6uh7jar
From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96
Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Content-Length: 0

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

<— SIP read from WS:189.232.221.54:56882 —>
INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 INVITE
Contact: sip:u18vq2jc@1lhjsfc8k672.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: 2238

v=0
o=- 7089255009147564118 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:57129 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:0s4bNH9Jo66GKXst
a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M
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:112670393 cname:rahmGKoZTz8GDKNr
a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a
a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a
<------------->
— (13 headers 47 lines) —
Using INVITE request as basis request - 73c6ks5m99l72uddthv6
Found peer ‘8001’ for ‘8001’ from 189.232.221.54:56882

<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as52137f59
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 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=“webrtc.no-ip.org”, nonce="04f22c86"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘73c6ks5m99l72uddthv6’ in 17472 ms (Method: INVITE)

<— SIP read from WS:189.232.221.54:56882 —>
ACK sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450
To: sip:1000@webrtc.no-ip.org;tag=as52137f59
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 ACK

<------------->
— (6 headers 0 lines) —

<— SIP read from WS:189.232.221.54:56882 —>
INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“04f22c86”, uri="sip:1000@webrtc.no-ip.org", response="651e8183d2aa4335c625fbe88131efe7"
Contact: sip:u18vq2jc@1lhjsfc8k672.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: 2238

v=0
o=- 7089255009147564118 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:57129 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:0s4bNH9Jo66GKXst
a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M
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:112670393 cname:rahmGKoZTz8GDKNr
a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a
a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a
<------------->
— (14 headers 47 lines) —
Using INVITE request as basis request - 73c6ks5m99l72uddthv6
Found peer ‘8001’ for ‘8001’ from 189.232.221.54:56882
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.101.103:57129
Looking for 1000 in default (domain webrtc.no-ip.org)
list_route: hop: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Length: 0

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

OKAudio is at 14566

0KAdding codec 100003 (ulaw) to SDP

0KAdding non-codec 0x1 (telephone-event) to SDP

0K
<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Type: application/sdp
Content-Length: 321

v=0
o=root 385143318 385143318 IN IP4 192.168.192.7
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.192.7
t=0 0
m=audio 14566 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:uSNE9caBZA2PPo6wPPWYkco5RrYdwU+jZyWdMXfE

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

<— SIP read from WS:189.232.221.54:56882 —>
ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5964741
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from WS:189.232.221.54:56882 —>
BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6707 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Scheduling destruction of SIP dialog ‘73c6ks5m99l72uddthv6’ in 17472 ms (Method: BYE)

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6707 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

[/code]

JavaSCRIPTDebug.LOG

[code]GET http://tryit.jssip.net/js/custom.js 404 (Not Found) tryit.jssip.net/:22
event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery-1.6.1.min.js:17
JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783
• via_host: “1lhjsfc8k672.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:webrtc.no-ip.org 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:74.125.132.127:19302”] 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:8001@webrtc.no-ip.org jssip-devel.js:5788
• ws_servers: [{“ws_uri”:“ws://webrtc.no-ip.org:8088/ws”,“sip_uri”:“sip:webrtc.no-ip.org:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794
• display_name: “G” jssip-devel.js:5794
• instance_id: “a499f638-557f-45c2-87ab-52feb7df49b1” jssip-devel.js:5794
• jssip_id: “73c6k” jssip-devel.js:5794
• hostport_params: “webrtc.no-ip.org” jssip-devel.js:5794
• authorization_user: “8001” 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://webrtc.no-ip.org:8088/ws jssip-devel.js:568
JsSIP | TRANSPORT | WebSocket ws://webrtc.no-ip.org: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:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
Call-ID: 1jtu7lhr09rhmilv2k6sgu
CSeq: 81 REGISTER
Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;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 1lhjsfc8k672.invalid;branch=z9hG4bK8958760;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9
Call-ID: 1jtu7lhr09rhmilv2k6sgu
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=“webrtc.no-ip.org”, nonce="5a4ecdb4"
Content-Length: 0

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

REGISTER sip:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
Call-ID: 1jtu7lhr09rhmilv2k6sgu
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“5a4ecdb4”, uri=“sip:webrtc.no-ip.org”, response="8df3a2d05686d9bc7bc2b856bbe5fba0"
Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;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:

OPTIONS sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34
Max-Forwards: 70
From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96
To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws
Contact: sip:8001@192.168.192.7:5060;transport=WS
Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Thu, 24 Apr 2014 20:58:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34
To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;tag=p9g6uh7jar
From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96
Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Content-Length: 0

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

SIP/2.0 200 OK
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7
To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9
Call-ID: 1jtu7lhr09rhmilv2k6sgu
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:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;expires=600
Date: Thu, 24 Apr 2014 20:58:57 GMT
Content-Length: 0

jssip-devel.js:686
JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK07255c34 jssip-devel.js:2094
Registered init.js:428
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:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 INVITE
Contact: sip:u18vq2jc@1lhjsfc8k672.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: 2238

v=0
o=- 7089255009147564118 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:57129 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:0s4bNH9Jo66GKXst
a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M
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:112670393 cname:rahmGKoZTz8GDKNr
a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a
a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a

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

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as52137f59
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 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=“webrtc.no-ip.org”, nonce="04f22c86"
Content-Length: 0

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

ACK sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450
To: sip:1000@webrtc.no-ip.org;tag=as52137f59
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6705 ACK

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

INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“04f22c86”, uri="sip:1000@webrtc.no-ip.org", response="651e8183d2aa4335c625fbe88131efe7"
Contact: sip:u18vq2jc@1lhjsfc8k672.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: 2238

v=0
o=- 7089255009147564118 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:57129 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:0s4bNH9Jo66GKXst
a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M
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:112670393 cname:rahmGKoZTz8GDKNr
a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a
a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm
a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a

jssip-devel.js:519
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK5326450 jssip-devel.js:1969
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;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 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Type: application/sdp
Content-Length: 321

v=0
o=root 385143318 385143318 IN IP4 192.168.192.7
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.192.7
t=0 0
m=audio 14566 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:uSNE9caBZA2PPo6wPPWYkco5RrYdwU+jZyWdMXfE

jssip-devel.js:686
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-devel.js:2546
Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd. jssip-devel.js:4544
JsSIP | TRANSPORT | sending WebSocket message:

ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5964741
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6706 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

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

BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6707 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | RTC SESSION | closing INVITE session 73c6ks5m99l72uddthv61s29gab6p9 jssip-devel.js:4248
JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424
JsSIP | DIALOG | dialog 73c6ks5m99l72uddthv61s29gab6p9as37deb063 deleted jssip-devel.js:2566
JsSIP | EVENT EMITTER | emitting event failed jssip-devel.js:187
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9
To: sip:1000@webrtc.no-ip.org;tag=as37deb063
Call-ID: 73c6ks5m99l72uddthv6
CSeq: 6707 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
[/code]

4

Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd. jssip-devel.js:4544 JsSIP | TRANSPORT | sending WebSocket message:

Seems like your asterisk is not compiled with ICE Support install libuuid/uuid and libuuid-devel/uuid-devel and recompile asterisk then try again.

5

Thanks for the tip, it solved the connection problem, but still can not get audio on the web page.
Here are the new log files.

SIP.log

[code]<— SIP read from WS:189.232.221.54:59385 —>
REGISTER sip:XXXX.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@XXXX.no-ip.org;tag=lftr2g6uci
Call-ID: m94e59t1sqbl2pobami4no
CSeq: 81 REGISTER
Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

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

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8
Call-ID: m94e59t1sqbl2pobami4no
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=“webrtc.no-ip.org”, nonce="15ee906d"
Content-Length: 0

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

<— SIP read from WS:189.232.221.54:59385 —>
REGISTER sip:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
Call-ID: m94e59t1sqbl2pobami4no
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“15ee906d”, uri=“sip:webrtc.no-ip.org”, response="332f925df91f25dbfbaf82267a6c87d1"
Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

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

Reliably Transmitting (no NAT) to 189.232.221.54:59385:
OPTIONS sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557
Max-Forwards: 70
From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738
To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws
Contact: sip:8001@192.168.192.7:5060;transport=WS
Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Fri, 25 Apr 2014 01:44:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8
Call-ID: m94e59t1sqbl2pobami4no
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:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;expires=600
Date: Fri, 25 Apr 2014 01:44:11 GMT
Content-Length: 0

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

Scheduling destruction of SIP dialog ‘m94e59t1sqbl2pobami4no’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:189.232.221.54:59385 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557
To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;tag=nljv5atrao
From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738
Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Content-Length: 0

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

<— SIP read from WS:189.232.221.54:59385 —>
INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 INVITE
Contact: sip:06sfurc9@7h1b1f5fc8gi.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: 2242

v=0
o=- 3652154159910496246 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:54280 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:eNWxE4gTTqnCVyMu
a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8
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:3951139034 cname:1GEOwdZBp2a0BbWt
a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
<------------->
— (13 headers 47 lines) —
Using INVITE request as basis request - cgu3s60etphsvr40mneg
Found peer ‘8001’ for ‘8001’ from 189.232.221.54:59385

<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as0464893c
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 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=“webrtc.no-ip.org”, nonce="12671f92"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘cgu3s60etphsvr40mneg’ in 15552 ms (Method: INVITE)

<— SIP read from WS:189.232.221.54:59385 —>
ACK sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051
To: sip:1000@webrtc.no-ip.org;tag=as0464893c
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 ACK

<------------->
— (6 headers 0 lines) —

<— SIP read from WS:189.232.221.54:59385 —>
INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“12671f92”, uri="sip:1000@webrtc.no-ip.org", response="63ef359a7befb51ebef6a306a7262d82"
Contact: sip:06sfurc9@7h1b1f5fc8gi.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: 2242

v=0
o=- 3652154159910496246 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:54280 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:eNWxE4gTTqnCVyMu
a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8
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:3951139034 cname:1GEOwdZBp2a0BbWt
a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
<------------->
— (14 headers 47 lines) —
Using INVITE request as basis request - cgu3s60etphsvr40mneg
Found peer ‘8001’ for ‘8001’ from 189.232.221.54:59385

Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126

Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)

Peer audio RTP is at port 192.168.101.103:54280

Looking for 1000 in default (domain webrtc.no-ip.org)

list_route: hop: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Length: 0

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

Audio is at 16094

Adding codec 100003 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Type: application/sdp
Content-Length: 827

v=0
o=root 1916711608 1916711608 IN IP4 192.168.192.7
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.192.7
t=0 0
m=audio 16094 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:508b3b3a55b6364115e87667717fe8b3
a=ice-pwd:12147891493792b1612c66d02e20cfd0
a=candidate:Hc0a8c007 1 UDP 2130706431 192.168.192.7 16094 typ host
a=candidate:Hc0a80103 1 UDP 2130706431 192.168.1.3 16094 typ host
a=candidate:Sc953b8ef 1 UDP 1694498815 201.83.184.239 10332 typ srflx
a=candidate:Hc0a8c007 2 UDP 2130706430 192.168.192.7 16095 typ host
a=candidate:Hc0a80103 2 UDP 2130706430 192.168.1.3 16095 typ host
a=candidate:Sc953b8ef 2 UDP 1694498814 201.83.184.239 10332 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:5HrqW3GMP29G3i/zeH2c6B1DTbOS0UNdeeuY6i/f

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

<— SIP read from WS:189.232.221.54:59385 —>
ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3660144
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

<------------->
— (10 headers 0 lines) —

Really destroying SIP dialog ‘m94e59t1sqbl2pobami4no’ Method: REGISTER

<— SIP read from WS:189.232.221.54:59385 —>
BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5921 BYE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘cgu3s60etphsvr40mneg’ in 15552 ms (Method: BYE)

<— Transmitting (no NAT) to 189.232.221.54:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5921 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

Really destroying SIP dialog ‘cgu3s60etphsvr40mneg’ Method: BYE

[/code]

JavaScriptDebug.log

[code]GET http://tryit.jssip.net/js/custom.js 404 (Not Found) tryit.jssip.net/:22
event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery-1.6.1.min.js:17
JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783
• via_host: “7h1b1f5fc8gi.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:webrtc.no-ip.org 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:74.125.132.127:19302”] 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:8001@webrtc.no-ip.org jssip-devel.js:5788
• ws_servers: [{“ws_uri”:“ws://webrtc.no-ip.org:8088/ws”,“sip_uri”:“sip:webrtc.no-ip.org:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794
• display_name: “G” jssip-devel.js:5794
• instance_id: “1b399baa-f209-4868-9e1e-490fb50fb0ea” jssip-devel.js:5794
• jssip_id: “cgu3s” jssip-devel.js:5794
• hostport_params: “webrtc.no-ip.org” jssip-devel.js:5794
• authorization_user: “8001” 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://webrtc.no-ip.org:8088/ws jssip-devel.js:568
JsSIP | TRANSPORT | WebSocket ws://webrtc.no-ip.org: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:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
Call-ID: m94e59t1sqbl2pobami4no
CSeq: 81 REGISTER
Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;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 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8
Call-ID: m94e59t1sqbl2pobami4no
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=“webrtc.no-ip.org”, nonce="15ee906d"
Content-Length: 0

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

REGISTER sip:webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886
Max-Forwards: 69
To: sip:8001@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
Call-ID: m94e59t1sqbl2pobami4no
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“15ee906d”, uri=“sip:webrtc.no-ip.org”, response="332f925df91f25dbfbaf82267a6c87d1"
Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;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:

OPTIONS sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557
Max-Forwards: 70
From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738
To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws
Contact: sip:8001@192.168.192.7:5060;transport=WS
Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Fri, 25 Apr 2014 01:44:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557
To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;tag=nljv5atrao
From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738
Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Content-Length: 0

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

SIP/2.0 200 OK
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci
To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8
Call-ID: m94e59t1sqbl2pobami4no
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:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;expires=600
Date: Fri, 25 Apr 2014 01:44:11 GMT
Content-Length: 0

jssip-devel.js:686
JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK56a57557 jssip-devel.js:2094
Registered init.js:428
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:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
jssip-devel.js:3401
JsSIP | TRANSPORT | sending WebSocket message:

INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 INVITE
Contact: sip:06sfurc9@7h1b1f5fc8gi.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: 2242

v=0
o=- 3652154159910496246 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:54280 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:eNWxE4gTTqnCVyMu
a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8
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:3951139034 cname:1GEOwdZBp2a0BbWt
a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435

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

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as0464893c
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 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=“webrtc.no-ip.org”, nonce="12671f92"
Content-Length: 0

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

ACK sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051
To: sip:1000@webrtc.no-ip.org;tag=as0464893c
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 ACK

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

INVITE sip:1000@webrtc.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Authorization: Digest algorithm=MD5, username=“8001”, realm=“webrtc.no-ip.org”, nonce=“12671f92”, uri="sip:1000@webrtc.no-ip.org", response="63ef359a7befb51ebef6a306a7262d82"
Contact: sip:06sfurc9@7h1b1f5fc8gi.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: 2242

v=0
o=- 3652154159910496246 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:54280 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:eNWxE4gTTqnCVyMu
a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
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:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8
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:3951139034 cname:1GEOwdZBp2a0BbWt
a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435

jssip-devel.js:519
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK9976051 jssip-devel.js:1969
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;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 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1000@192.168.192.7:5060;transport=WS
Content-Type: application/sdp
Content-Length: 827

v=0
o=root 1916711608 1916711608 IN IP4 192.168.192.7
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.192.7
t=0 0
m=audio 16094 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:508b3b3a55b6364115e87667717fe8b3
a=ice-pwd:12147891493792b1612c66d02e20cfd0
a=candidate:Hc0a8c007 1 UDP 2130706431 192.168.192.7 16094 typ host
a=candidate:Hc0a80103 1 UDP 2130706431 192.168.1.3 16094 typ host
a=candidate:Sc953b8ef 1 UDP 1694498815 201.83.184.239 10332 typ srflx
a=candidate:Hc0a8c007 2 UDP 2130706430 192.168.192.7 16095 typ host
a=candidate:Hc0a80103 2 UDP 2130706430 192.168.1.3 16095 typ host
a=candidate:Sc953b8ef 2 UDP 1694498814 201.83.184.239 10332 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:5HrqW3GMP29G3i/zeH2c6B1DTbOS0UNdeeuY6i/f

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:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3660144
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5920 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 | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK7110183 jssip-devel.js:1960
JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK7110183 jssip-devel.js:1949
JsSIP | RTC SESSION | terminating RTCSession jssip-devel.js:3790
JsSIP | TRANSPORT | sending WebSocket message:

BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967
Max-Forwards: 69
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5921 BYE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0

jssip-devel.js:519
JsSIP | RTC SESSION | closing INVITE session cgu3s60etphsvr40mnegi6i4nniqbv jssip-devel.js:4248
JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424
JsSIP | DIALOG | dialog cgu3s60etphsvr40mnegi6i4nniqbvas40e79ab8 deleted jssip-devel.js:2566
JsSIP | EVENT EMITTER | emitting event ended jssip-devel.js:187
JsSIP | TRANSPORT | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967;received=189.232.221.54
From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv
To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5921 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[/code]

6

The SDP is using a diferent IP, the signaling send the INVITE from 179.213.21.254 and the SDP use: 192.168.101.103. This issue is related to your NAT settings or ice/stun servers.

7

Thank you very much!

In my sip.conf i used

nat=auto_force_rport,auto_comedia

I just alter to

nat=force_rport,comedia

and the audio start to work.

Thanks

Geraldo