Help webRTC sipml5

angelope · March 9, 2015, 4:02pm

Hello,

deb 6.0.7 on VMware workstation 10
Asterisk: 11.8

I’ve been configuring webrtc following theses guides:

wiki.asterisk.org/wiki/display/ … ing+SIPML5

wiki.asterisk.org/wiki/display/ … figuration

I’ve follow everystep and everything looks ok during installation, also I can register remote client to my Asterisk but i get the following error when I try to make a call

No DTLS-SRTP support present on engine for RTP instance

It’s a strange error becouse modules are loaded correctly

Asterisk11*CLI> module show like rtp
Module Description Use Count
chan_multicast_rtp.so Multicast RTP Paging Channel 0
res_rtp_asterisk.so Asterisk RTP Stack 1
res_rtp_multicast.so Multicast RTP Engine 0
res_srtp.so Secure RTP (SRTP) 0
4 modules loaded

I’ve been workng with this for two days but no results…

Any help would be wellcome

thanks in advance

navaismo · March 9, 2015, 5:33pm

Use the latest version of trunk 11 and make sure you have configured correctly the DTLS-SRTP part in your sip.conf.

angelope · March 9, 2015, 6:31pm

Hi,

Already tested on Asterisk 11.16, with no results, configs looks ok like in the guide…

Could be a virtualization problems

Thx again

navaismo · March 9, 2015, 6:33pm

No your issue is with your config, many users already tested on VMs

angelope · March 9, 2015, 8:17pm

These are my configs
sip .conf


[general]



udpbindaddr=0.0.0.0:5060
realm=xx.xx.xx.xx ;replace with your Asterisk server public IP address or host
transport=udp,ws



[6001]
host=dynamic
secret=6001
context=local
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

rtp.conf

;
; RTP Configuration
;
[general]
;

;
rtpstart=10000
rtpend=20000
;
;
;rtpchecksums=no
;

; Whether to enable or disable ICE support. This option is disabled by default.
icesupport=true

stunaddr=stun.l.google.com:19302

http.conf

;       http://<server_ip>:<bindport>/static/docs/index.html
;
[general]
;
; Whether HTTP/HTTPS interface is enabled or not.  Default is no.
; This also affects manager/rawman/mxml access (see manager.conf)
;
enabled=yes
;
; Address to bind to, both for HTTP and HTTPS. You MUST specify
; a bindaddr in order for the HTTP server to run. There is no
; default value.
;
bindaddr=0.0.0.0
;
; Port to bind to for HTTP sessions (default is 8088)
;
bindport=8088

Any help would be really apriciate

navaismo · March 9, 2015, 8:24pm

As mentioned i n the sticky post of webrtc on this forum you need to provide basic debug logs to get futher help. Please read it and come back with the log of the failure.

angelope · March 9, 2015, 8:47pm

sip set debug on

[code]Connected to Asterisk 11.16.0 currently running on Asterisk11 (pid = 20990)
== WebSocket connection from ‘xx.xx.xxx.50:18638’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘6001’ at xx.xx.xxx.50:18638
Asterisk11CLI> sip set debug on
SIP Debugging enabled
Asterisk11CLI>

<— SIP read from UDP:192.168.1.100:45563 —>

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

<— SIP read from WS:xx.xx.xxx.50:18638 —>
INVITE sip:200@8xx.xx.xxx.50 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaBxW5IH8l527DQIm4lXDLocVoc6vTtyk;rport
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: sip:200@xx.xx.xxx.50
Contact: "Angel"sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language="en,fr"
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43140 INVITE
Content-Type: application/sdp
Content-Length: 2539
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 2262593349607579100 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn
m=audio 52853 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xxx.50
a=rtcp:52853 IN IP4 xx.xx.xxx.50
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 52853 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 52853 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 52854 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 52854 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 52855 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 52855 typ host generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xxx.50 52853 typ srflx raddr 192.168.1.100 rport 52853 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xxx.50 52853 typ srflx raddr 192.168.1.100 rport 52853 generation 0
a=ice-ufrag:8g2ge/nK3eRTs3As
a=ice-pwd:uvKDXrbIv7nUzwdDTS4PXmnv
a=ice-options:google-ice
a=fingerprint:sha-256 07:10:62:F8:8A:75:71:B7:E1:27:22:2A:7F:3F:E3:FE:18:2A:1B:AC:E7:95:FE:91:D1:27:04:B8:49:04:0B:30
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2359155316 cname:jctxCR/ysJg86DIe
a=ssrc:2359155316 msid:x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn 2e03bc41-7106-4fdf-aefd-8ed5dbded942
a=ssrc:2359155316 mslabel:x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn
a=ssrc:2359155316 label:2e03bc41-7106-4fdf-aefd-8ed5dbded942
<------------->
— (12 headers 49 lines) —
Using INVITE request as basis request - 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
Found peer ‘6001’ for ‘6001’ from xx.xx.xxx.50:18638

<— Reliably Transmitting (no NAT) to xx.xx.xxx.50:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaBxW5IH8l527DQIm4lXDLocVoc6vTtyk;rport;received=xx.xx.xxx.50
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: <sip:200@xx.xx.xxx.50;tag=as4ebc7c65
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43140 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“xx.xx.xxx.50”, nonce="5f9a4f3b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f’ in 32000 ms (Method: INVITE)

<— SIP read from WS:xx.xx.xxx.50:18638 —>
ACK sip:200@xx.xx.xxx.50 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaBxW5IH8l527DQIm4lXDLocVoc6vTtyk;rport
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: sip:200@xx.xx.xxx.50;tag=as4ebc7c65
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43140 ACK
Content-Length: 0
Max-Forwards: 70

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

<— SIP read from WS:xx.xx.xxx.50:18638 —>
INVITE sip:200@xx.xx.xxx.50 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvTs62gWPdodQCFdUoCKc9oWWacm2sTub;rport
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: sip:200@xx.xx.xxx.50
Contact: "Angel"sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language="en,fr"
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43141 INVITE
Content-Type: application/sdp
Content-Length: 2539
Max-Forwards: 70
Authorization: Digest username=“6001”,realm=“xx.xx.xxx.50”,nonce=“5f9a4f3b”,uri=“sip:200@xx.xx.xxx.50”,response=“81335f4ea7bd2cf8786457e6947f50e6”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 2262593349607579100 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn
m=audio 52853 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xxx.50
a=rtcp:52853 IN IP4 xx.xx.xxx.50
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 52853 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 52853 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 52854 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 52854 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 52855 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 52855 typ host generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xxx.50 52853 typ srflx raddr 192.168.1.100 rport 52853 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xxx.50 52853 typ srflx raddr 192.168.1.100 rport 52853 generation 0
a=ice-ufrag:8g2ge/nK3eRTs3As
a=ice-pwd:uvKDXrbIv7nUzwdDTS4PXmnv
a=ice-options:google-ice
a=fingerprint:sha-256 07:10:62:F8:8A:75:71:B7:E1:27:22:2A:7F:3F:E3:FE:18:2A:1B:AC:E7:95:FE:91:D1:27:04:B8:49:04:0B:30
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2359155316 cname:jctxCR/ysJg86DIe
a=ssrc:2359155316 msid:x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn 2e03bc41-7106-4fdf-aefd-8ed5dbded942
a=ssrc:2359155316 mslabel:x1BPenJDwbFjKqmZRD8JdqcL0qE4RbldX2vn
a=ssrc:2359155316 label:2e03bc41-7106-4fdf-aefd-8ed5dbded942
<------------->
— (13 headers 49 lines) —
Using INVITE request as basis request - 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
Found peer ‘6001’ for ‘6001’ from xx.xx.xxx.50:18638
[Mar 9 21:41:52] ERROR[21067][C-00000004]: chan_sip.c:5702 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance ‘0x33a25a8’, was it compiled with support for it?
[Mar 9 21:41:52] NOTICE[21067][C-00000004]: chan_sip.c:25666 handle_request_invite: Failed to authenticate device "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK

<— Reliably Transmitting (no NAT) to xx.xx.xxx.50:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvTs62gWPdodQCFdUoCKc9oWWacm2sTub;rport;received=xx.xx.xxx.50
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: sip:200@xx.xx.xxx.50;tag=as4ebc7c65
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43141 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f’ in 32000 ms (Method: INVITE)

<— SIP read from WS:xx.xx.xxx.50:18638 —>
ACK sip:200@xx.xx.xxx.50SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvTs62gWPdodQCFdUoCKc9oWWacm2sTub;rport
From: "Angel"sip:6001@xx.xx.xxx.50;tag=5nyhARoCPhRzeLXTzwLK
To: sip:200@xx.xx.xxx.50;tag=as4ebc7c65
Call-ID: 9a99d7ae-428b-e723-cf74-bf8a8e5d9f3f
CSeq: 43141 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘5bf91fa1-deae-88d8-3a94-b7e66dcf0c32’ Method: REGISTER
Asterisk11*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[/code]

angelope · March 9, 2015, 8:49pm

Thanks again

navaismo · March 10, 2015, 2:31pm

Read this and try again: viewtopic.php?f=1&t=91007

If you already installed all the dependencies then your libraries are too old and you need to upgrade it.

angelope · March 12, 2015, 6:21pm

Hi;

Thanks for the response, that did the trick, it looks like a problem with my debian repos, us repo added, then apt-get update, apt-get upgrade, and recompiliing Asterisk.

No I´ve the “famous” Called with SDP without ice-ufrag and ice-pwd issue.

I´ve read all posts from Asterisk support, but I can´t find my mistake.

All packages look find:

dpkg -l | grep uuid

ii libossp-uuid16 1.6.2-1 OSSP uuid ISO-C and C++ - shared library
ii libuuid-perl 0.02-4 Perl extension for using UUID interfaces as defined in e2fsprogs
ii libuuid1 2.17.2-9 Universally Unique ID library
ii uuid 1.6.2-1 the Universally Unique Identifier Command-Line Tool
ii uuid-dev 2.17.2-9 universally unique id library - headers and static libraries

Also correct ip address in sip debugs…

But reviewing my sip debug i found the following error:

<— Reliably Transmitting (NAT) to xx.xx.xx.150:15171 —>
SIP/2.0 401 unautorithed
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJ6U1ukGwHQlzv87aeI9n2N9PgFtdcJbJ;received=xx.xx.xx.150;rport=15171
From: "Angel"sip:6001@xx.xx.xx.150;tag=ULZ1xaZWO9xbpSJiVLby
To: sip:200@xx.xx.xx.150;tag=as64edcf1a
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1884 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“xx.xx.xx.150”, nonce="6d8e536f"
Content-Length: 0

Here is the full debug

Asterisk11*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from WS:xx.xx.xx.150:15171 --->
INVITE sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJ6U1ukGwHQlzv87aeI9n2N9PgFtdcJbJ;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1884 INVITE
Content-Type: application/sdp
Content-Length: 2539
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 4558366527426015000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo
m=audio 58798 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:58798 IN IP4 xx.xx.xx.150
a=candidate:186199869 1 udp 2122260223 192.168.1.101 58798 typ host generation 0
a=candidate:186199869 2 udp 2122260223 192.168.1.101 58798 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 58799 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 58799 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 58800 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 58800 typ host generation 0
a=candidate:1167774669 1 tcp 1518280447 192.168.1.101 0 typ host tcptype active generation 0
a=candidate:1167774669 2 tcp 1518280447 192.168.1.101 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2320574857 1 udp 1686052607 xx.xx.xx.150 58798 typ srflx raddr 192.168.1.101 rport 58798 generation 0
a=candidate:2320574857 2 udp 1686052607 xx.xx.xx.150 58798 typ srflx raddr 192.168.1.101 rport 58798 generation 0
a=ice-ufrag:fJR6QDyp2AqN11T/
a=ice-pwd:NKRgrPcC45Xxp2YC2lr4T6u3
a=ice-options:google-ice
a=fingerprint:sha-256 F2:E9:97:27:D2:51:73:A7:13:0C:2E:E8:DC:11:56:F8:BB:87:69:A6:59:63:EC:81:88:45:9A:10:23:25:20:B1
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:3941431346 cname:Nlgf/1d7OvPzH6pA
a=ssrc:3941431346 msid:ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo 281356e0-f9df-4979-ba26-d6d3f0db2021
a=ssrc:3941431346 mslabel:ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo
a=ssrc:3941431346 label:281356e0-f9df-4979-ba26-d6d3f0db2021
<------------->
--- (12 headers 49 lines) ---
Using INVITE request as basis request - e11a9516-e183-70eb-0083-41649fc9ddb2
Found peer '6001' for '6001' from xx.xx.xx.150:15171

<--- Reliably Transmitting (NAT) to xx.xx.xx.150:15171 --->
SIP/2.0 401 unautorithed
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJ6U1ukGwHQlzv87aeI9n2N9PgFtdcJbJ;received=xx.xx.xx.150;rport=15171
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>;tag=as64edcf1a
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1884 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="xx.xx.xx.150", nonce="6d8e536f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e11a9516-e183-70eb-0083-41649fc9ddb2' in 32000 ms (Method: INVITE)

<--- SIP read from WS:xx.xx.xx.150:15171 --->
ACK sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJ6U1ukGwHQlzv87aeI9n2N9PgFtdcJbJ;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>;tag=as64edcf1a
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1884 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:xx.xx.xx.150:15171 --->
INVITE sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBCXUEJMLItmKrXtvffQnqL6oXFc2CtvJ;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1885 INVITE
Content-Type: application/sdp
Content-Length: 2539
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="6d8e536f",uri="sip:200@xx.xx.xx.150",response="9a58f0111b0c49d9ba530dec303775b5",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 4558366527426015000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo
m=audio 58798 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:58798 IN IP4 xx.xx.xx.150
a=candidate:186199869 1 udp 2122260223 192.168.1.101 58798 typ host generation 0
a=candidate:186199869 2 udp 2122260223 192.168.1.101 58798 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 58799 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 58799 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 58800 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 58800 typ host generation 0
a=candidate:1167774669 1 tcp 1518280447 192.168.1.101 0 typ host tcptype active generation 0
a=candidate:1167774669 2 tcp 1518280447 192.168.1.101 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2320574857 1 udp 1686052607 xx.xx.xx.150 58798 typ srflx raddr 192.168.1.101 rport 58798 generation 0
a=candidate:2320574857 2 udp 1686052607 xx.xx.xx.150 58798 typ srflx raddr 192.168.1.101 rport 58798 generation 0
a=ice-ufrag:fJR6QDyp2AqN11T/
a=ice-pwd:NKRgrPcC45Xxp2YC2lr4T6u3
a=ice-options:google-ice
a=fingerprint:sha-256 F2:E9:97:27:D2:51:73:A7:13:0C:2E:E8:DC:11:56:F8:BB:87:69:A6:59:63:EC:81:88:45:9A:10:23:25:20:B1
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:3941431346 cname:Nlgf/1d7OvPzH6pA
a=ssrc:3941431346 msid:ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo 281356e0-f9df-4979-ba26-d6d3f0db2021
a=ssrc:3941431346 mslabel:ZjTOFc30LASXX4FhBdmbP6vwnnO5amntP5qo
a=ssrc:3941431346 label:281356e0-f9df-4979-ba26-d6d3f0db2021
<------------->
--- (13 headers 49 lines) ---
Using INVITE request as basis request - e11a9516-e183-70eb-0083-41649fc9ddb2
Found peer '6001' for '6001' from xx.xx.xx.150:15171
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 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 G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/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 xx.xx.xx.150:58798
Looking for 200 in local (domain xx.xx.xx.150)
list_route: hop: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>

<--- Transmitting (NAT) to xx.xx.xx.150:15171 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBCXUEJMLItmKrXtvffQnqL6oXFc2CtvJ;received=xx.xx.xx.150;rport=15171
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1885 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@xx.xx.xx.150:0;transport=WS>
Content-Length: 0


<------------>
    -- Executing [200@local:1] Answer("SIP/6001-00000001", "") in new stack
Audio is at 11118
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to xx.xx.xx.150:15171 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBCXUEJMLItmKrXtvffQnqL6oXFc2CtvJ;received=xx.xx.xx.150;rport=15171
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>;tag=as32083b64
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1885 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@xx.xx.xx.150:0;transport=WS>
Content-Type: application/sdp
Content-Length: 393

v=0
o=root 1075372712 1075372712 IN IP4 xx.xx.xx.150
s=Asterisk PBX 11.16.0
c=IN IP4 xx.xx.xx.150
t=0 0
m=audio 11118 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 89:5F:52:19:9D:B3:EF:89:6E:22:F7:C7:AB:DC:F5:AB:F6:B4:F1:07:F2:A7:A6:5D:93:EC:34:B0:8D:E1:20:BC
a=sendrecv

<------------>
    -- Executing [200@local:2] Playback("SIP/6001-00000001", "hello-world") in new stack
    -- <SIP/6001-00000001> Playing 'hello-world.gsm' (language 'en')

<--- SIP read from WS:xx.xx.xx.150:15171 --->
ACK sip:200@xx.xx.xx.150;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK3AnK9hxl5acaWNqKVeIu;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
To: <sip:200@xx.xx.xx.150>;tag=as32083b64
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 1885 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="6d8e536f",uri="sip:200@xx.xx.xx.150;transport=WS",response="59dafa36878a3d8f2e878e0b6446f1bd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (12 headers 0 lines) ---
    -- Executing [200@local:3] Hangup("SIP/6001-00000001", "") in new stack
  == Spawn extension (local, 200, 3) exited non-zero on 'SIP/6001-00000001'
Scheduling destruction of SIP dialog 'e11a9516-e183-70eb-0083-41649fc9ddb2' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (NAT) to xx.xx.xx.150:15171:
BYE sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS xx.xx.xx.150:0;branch=z9hG4bK381bca57;rport
Max-Forwards: 70
From: <sip:200@xx.xx.xx.150>;tag=as32083b64
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.16.0
Proxy-Authorization: Digest username="6001", realm="xx.xx.xx.150", algorithm=MD5, uri="sip:xx.xx.xx.150", nonce="6d8e536f", response="c5d62e7435f11a7b2996310b8b855fdc"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:xx.xx.xx.150:15171 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS xx.xx.xx.150;rport;branch=z9hG4bK381bca57
From: <sip:200@xx.xx.xx.150>;tag=as32083b64
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=ULZ1xaZWO9xbpSJiVLby
Contact: <sip:6001@df7jal23ls0d.invalid;transport=ws>
Call-ID: e11a9516-e183-70eb-0083-41649fc9ddb2
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'e11a9516-e183-70eb-0083-41649fc9ddb2' Method: INVITE

<--- SIP read from UDP:192.168.1.101:63300 --->


<------------->
Asterisk11*CLI>
Asterisk11*CLI>
Asterisk11*CLI> sip set debug off

Any suggestion would be appreciated

Thanks in advance

angelope · March 12, 2015, 7:38pm

In SDP it looks like ice info is ok

a=ice-ufrag:fJR6QDyp2AqN11T/ a=ice-pwd:NKRgrPcC45Xxp2YC2lr4T6u3 a=ice-options:google-ice

Correct me if I´m wrong

navaismo · March 12, 2015, 8:53pm

But that’s from the Browser, Asterisk must send it too in the answer. If I recall the issue is because the engine wasn’t compiled with ice support, check if the res_rtp module has it by runing the ldd command.

Try using the lastest version of your trunk and make a distclean on the source folder then compile all from scratch again.

angelope · March 13, 2015, 4:49pm

IMO ldd looks good too…

ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so linux-vdso.so.1 => (0x00007fff671ff000) libuuid.so.1 => /lib/x86_64-linux-gnu/libuuid.so.1 (0x00007faa91026000) libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007faa90e0a000) libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007faa90a7d000) /lib64/ld-linux-x86-64.so.2 (0x00007faa9144e000)

Probably Asterisk was compiled with old libraries…

Now I’m building from scracth, (so and Asterisk) I´ll let you know

angelope · March 14, 2015, 4:31pm

Finally I managed to make this work with debian 6.0.7 and Asterisk 11.16, I added some more repos and then apt-get update apt-get upgrade and also apt-get dist-upgrade

I recompile Asterisk too

Calls are completed and have audio, but sometimes sipml5 client get disconected…

Also I´ve this error at Chrome console

Not implemented

SIPml-api.js?svn=222:1 tsk_utils_log_errorSIPml-api.js?svn=222:3 tsip_dialog_layer.handle_incoming_messageSIPml-api.js?svn=222:3 tsip_transport_layer.handle_incoming_messageSIPml-api.js?svn=222:3 __tsip_transport_ws_onmessage[/code]


This is the complete sip debug

[code]debian*CLI>
debian*CLI>
debian*CLI>
debian*CLI>
Really destroying SIP dialog '4d906879-eee8-3a47-32a9-3e7a3b8473cd' Method: REGISTER

<--- SIP read from UDP:192.168.1.100:46694 --->


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

<--- SIP read from WS:xx.xx.xx.150:10048 --->
INVITE sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdvOCRTT2kkoorBRU0pZVPuMDXylvur2K;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28791 INVITE
Content-Type: application/sdp
Content-Length: 2535
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 6736745922331888000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu
m=audio 63801 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:63801 IN IP4 xx.xx.xx.150
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 63801 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 63801 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 63802 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 63802 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 63803 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 63803 typ host generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xx.150 63801 typ srflx raddr 192.168.1.100 rport 63801 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xx.150 63801 typ srflx raddr 192.168.1.100 rport 63801 generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=ice-ufrag:PtKll7dhr/yzikf7
a=ice-pwd:lUMVyI8sRUOZnrVvcfx8r18Q
a=ice-options:google-ice
a=fingerprint:sha-256 F2:38:FC:8F:09:C5:A7:85:18:8D:BC:E7:BD:51:BD:D1:4B:37:E1:37:83:42:77:4A:92:25:25:FE:97:7E:9C:91
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:560846480 cname:c7iGalH8Mm6sEryB
a=ssrc:560846480 msid:AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu 9f5e420f-d352-4c36-93c5-fe8f12caa5e5
a=ssrc:560846480 mslabel:AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu
a=ssrc:560846480 label:9f5e420f-d352-4c36-93c5-fe8f12caa5e5
<------------->
--- (12 headers 49 lines) ---
Using INVITE request as basis request - 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
Found peer '6001' for '6001' from xx.xx.xx.150:10048

<--- Reliably Transmitting (NAT) to xx.xx.xx.150:10048 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdvOCRTT2kkoorBRU0pZVPuMDXylvur2K;received=xx.xx.xx.150;rport=10048
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>;tag=as57b8845a
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28791 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="xx.xx.xx.150", nonce="5825e0a1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89' in 49152 ms (Method: INVITE)

<--- SIP read from WS:xx.xx.xx.150:10048 --->
ACK sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdvOCRTT2kkoorBRU0pZVPuMDXylvur2K;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>;tag=as57b8845a
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28791 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:xx.xx.xx.150:10048 --->
INVITE sip:200@xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBVx9e4K4HpNX7TQ10ewfmpTo3WD24fg0;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28792 INVITE
Content-Type: application/sdp
Content-Length: 2535
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="5825e0a1",uri="sip:200@xx.xx.xx.150",response="87037b69155f8539d827a738e1ecec70",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 6736745922331888000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu
m=audio 63801 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:63801 IN IP4 xx.xx.xx.150
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 63801 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 63801 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 63802 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 63802 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 63803 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 63803 typ host generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xx.150 63801 typ srflx raddr 192.168.1.100 rport 63801 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xx.150 63801 typ srflx raddr 192.168.1.100 rport 63801 generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=ice-ufrag:PtKll7dhr/yzikf7
a=ice-pwd:lUMVyI8sRUOZnrVvcfx8r18Q
a=ice-options:google-ice
a=fingerprint:sha-256 F2:38:FC:8F:09:C5:A7:85:18:8D:BC:E7:BD:51:BD:D1:4B:37:E1:37:83:42:77:4A:92:25:25:FE:97:7E:9C:91
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:560846480 cname:c7iGalH8Mm6sEryB
a=ssrc:560846480 msid:AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu 9f5e420f-d352-4c36-93c5-fe8f12caa5e5
a=ssrc:560846480 mslabel:AKuKIMYSJ2P1SIae4Qvy09mUrNID50pcwtMu
a=ssrc:560846480 label:9f5e420f-d352-4c36-93c5-fe8f12caa5e5
<------------->
--- (13 headers 49 lines) ---
Using INVITE request as basis request - 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
Found peer '6001' for '6001' from xx.xx.xx.150:10048
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 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 G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port xx.xx.xx.150:63801
Looking for 200 in local (domain xx.xx.xx.150)
list_route: hop: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>

<--- Transmitting (NAT) to xx.xx.xx.150:10048 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBVx9e4K4HpNX7TQ10ewfmpTo3WD24fg0;received=xx.xx.xx.150;rport=10048
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28792 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@xx.xx.xx.150:0;transport=WS>
Content-Length: 0


<------------>
    -- Executing [200@local:1] Answer("SIP/6001-00000009", "") in new stack
Audio is at 12128
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to xx.xx.xx.150:10048 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBVx9e4K4HpNX7TQ10ewfmpTo3WD24fg0;received=xx.xx.xx.150;rport=10048
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>;tag=as21d9d47a
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28792 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@xx.xx.xx.150:0;transport=WS>
Content-Type: application/sdp
Content-Length: 853

v=0
o=root 1549315709 1549315709 IN IP4 xx.xx.xx.150
s=Asterisk PBX 11.16.0
c=IN IP4 xx.xx.xx.150
t=0 0
m=audio 12128 UDP/TLS/RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:28b004b7796d3a6b274c71b70f5f8b1a
a=ice-pwd:3a1a84773265770e35f87e942f99178c
a=candidate:Hc0a80146 1 UDP 2130706431 192.168.1.70 12128 typ host
a=candidate:S50257096 1 UDP 1694498815 xx.xx.xx.150 12128 typ srflx raddr 192.168.1.70 rport 12128
a=candidate:Hc0a80146 2 UDP 2130706430 192.168.1.70 12129 typ host
a=candidate:S50257096 2 UDP 1694498814 xx.xx.xx.150 12129 typ srflx raddr 192.168.1.70 rport 12129
a=connection:new
a=setup:active
a=fingerprint:SHA-256 09:58:31:FD:C7:12:BA:77:67:9F:52:CC:5C:86:27:60:B1:65:68:20:73:3C:CF:6A:02:C3:37:D3:4A:89:46:EF
a=sendrecv

<------------>
    -- Executing [200@local:2] Playback("SIP/6001-00000009", "hello-world") in new stack
    -- <SIP/6001-00000009> Playing 'hello-world.gsm' (language 'en')

<--- SIP read from WS:xx.xx.xx.150:10048 --->
ACK sip:200@xx.xx.xx.150;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTsauhcef0tOmPlvZyHuM;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
To: <sip:200@xx.xx.xx.150>;tag=as21d9d47a
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 28792 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="5825e0a1",uri="sip:200@xx.xx.xx.150;transport=WS",response="9590674af5c9fb5f17526171f65c7475",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (12 headers 0 lines) ---
       > 0x7fa4a0024f80 -- Probation passed - setting RTP source address to 192.168.1.100:63801
    -- Executing [200@local:3] Hangup("SIP/6001-00000009", "") in new stack
  == Spawn extension (local, 200, 3) exited non-zero on 'SIP/6001-00000009'
Scheduling destruction of SIP dialog '5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89' in 49152 ms (Method: INVITE)
set_destination: Parsing <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (NAT) to xx.xx.xx.150:10048:
BYE sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS xx.xx.xx.150:0;branch=z9hG4bK1d85b70a;rport
Max-Forwards: 70
From: <sip:200@xx.xx.xx.150>;tag=as21d9d47a
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 102 BYE
User-Agent: Digital-Merge_UA
Proxy-Authorization: Digest username="6001", realm="xx.xx.xx.150", algorithm=MD5, uri="sip:xx.xx.xx.150", nonce="5825e0a1", response="9932160d5471c487422453fe1878f4d9"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:xx.xx.xx.150:10048 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS xx.xx.xx.150;rport;branch=z9hG4bK1d85b70a
From: <sip:200@xx.xx.xx.150>;tag=as21d9d47a
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=zFV2K8PSbovuG4gzgXKC
Contact: <sip:6001@df7jal23ls0d.invalid;transport=ws>
Call-ID: 5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '5ea38a20-2c6e-3c2e-a0e4-e97b7fbe6e89' Method: INVITE
debian*CLI> sip set debug off
SIP Debugging Disabled
  == Using SIP RTP CoS mark 5
[Mar 14 17:20:51] NOTICE[19180][C-0000000b]: chan_sip.c:25759 handle_request_invite: Call from '' (195.154.56.42:5071) to extension '00972592420831' rejected because extension not found in context 'guest'.

This is the registration of the sipml5 client, I get error 405 and 401

debian*CLI>
  == WebSocket connection from 'xx.xx.xx.150:10256' for protocol 'sip' accepted using version '13'

<--- SIP read from WS:xx.xx.xx.150:10256 --->
REGISTER sip:xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTqdDqKPOMRsxTFKANxHiZaBVf2fDATUT;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=OrVBhGJWJEyvIaMeq1UH
To: "Angel"<sip:6001@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7deb69e6-244f-daf1-8250-e04828adf68c
CSeq: 53830 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="",uri="sip:xx.xx.xx.150",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom
Supported: path

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

<--- Transmitting (NAT) to xx.xx.xx.150:10256 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTqdDqKPOMRsxTFKANxHiZaBVf2fDATUT;received=xx.xx.xx.150;rport=10256
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=OrVBhGJWJEyvIaMeq1UH
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=as134e89d0
Call-ID: 7deb69e6-244f-daf1-8250-e04828adf68c
CSeq: 53830 REGISTER
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="xx.xx.xx.150", nonce="5caccba1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7deb69e6-244f-daf1-8250-e04828adf68c' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:xx.xx.xx.150:10256 --->
REGISTER sip:xx.xx.xx.150 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrsQxAiU9KWb0AwtQ5iZt1mRzlBJmIncf;rport
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=OrVBhGJWJEyvIaMeq1UH
To: "Angel"<sip:6001@xx.xx.xx.150>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7deb69e6-244f-daf1-8250-e04828adf68c
CSeq: 53831 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="xx.xx.xx.150",nonce="5caccba1",uri="sip:xx.xx.xx.150",response="cd9ea2d7630a877040834dd28a87d962",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom
Supported: path

<------------->
--- (13 headers 0 lines) ---
    -- Registered SIP '6001' at xx.xx.xx.150:10256
Reliably Transmitting (NAT) to xx.xx.xx.150:10256:
OPTIONS sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS xx.xx.xx.150:0;branch=z9hG4bK6090b4b5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@xx.xx.xx.150:0>;tag=as653ef531
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:asterisk@xx.xx.xx.150:0;transport=WS>
Call-ID: 5a329e3b16450d345ca7fc7d0dcd8724@xx.xx.xx.150:0
CSeq: 102 OPTIONS
User-Agent: Digital-Merge_UA
Date: Sat, 14 Mar 2015 16:25:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to xx.xx.xx.150:10256 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrsQxAiU9KWb0AwtQ5iZt1mRzlBJmIncf;received=xx.xx.xx.150;rport=10256
From: "Angel"<sip:6001@xx.xx.xx.150>;tag=OrVBhGJWJEyvIaMeq1UH
To: "Angel"<sip:6001@xx.xx.xx.150>;tag=as134e89d0
Call-ID: 7deb69e6-244f-daf1-8250-e04828adf68c
CSeq: 53831 REGISTER
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Date: Sat, 14 Mar 2015 16:25:35 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7deb69e6-244f-daf1-8250-e04828adf68c' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:xx.xx.xx.150:10256 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS xx.xx.xx.150;rport;branch=z9hG4bK6090b4b5
From: "asterisk"<sip:asterisk@xx.xx.xx.150>;tag=as653ef531
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 5a329e3b16450d345ca7fc7d0dcd8724@xx.xx.xx.150:0
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5a329e3b16450d345ca7fc7d0dcd8724@xx.xx.xx.150:0' Method: OPTIONS
debian*CLI> sip set debug off
SIP Debugging Disabled[/code]

These are ldd and dkpg commands:

[code][root@debian:~]#dpkg -l | grep uuid
ii  libossp-uuid16                     1.6.2-1                      amd64        OSSP uuid ISO-C and C++ - shared library
ii  libuuid-perl                       0.05-1+b1                    amd64        Perl extension for using UUID interfaces as defined in e2fsprogs
ii  libuuid1:amd64                     2.25.2-5                     amd64        Universally Unique ID library
ii  uuid                               1.6.2-1.5+b1                 amd64        Universally Unique Identifier Command-Line Tool
ii  uuid-dev:amd64                     2.25.2-5                     amd64        universally unique id library - headers and static libraries
[root@debian:~]#
[root@debian:~]#
[root@debian:~]#ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so
        linux-vdso.so.1 (0x00007fffc27ff000)
        libuuid.so.1 => /lib/x86_64-linux-gnu/libuuid.so.1 (0x00007fabfec22000)
        libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007fabfe921000)
        libnsl.so.1 => /lib/x86_64-linux-gnu/libnsl.so.1 (0x00007fabfe708000)
        librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007fabfe500000)
        libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007fabfe2e3000)
        libcrypto.so.1.0.0 => /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x00007fabfdee8000)
        libssl.so.1.0.0 => /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x00007fabfdc88000)
        libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fabfd8df000)
        /lib64/ld-linux-x86-64.so.2 (0x00007fabff088000)
        libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007fabfd6da000)
[root@debian:~]#

Any suggestions?

Thanks

navaismo · March 14, 2015, 6:14pm

The not implemented is not an error is the response for the option message sent by asterisk. Are you really using NAT? I saw that you are using y configs so first of all check if you really need NAT then check the codecs, the other log show that the extension dialed does not exist in the context so verify your dialplan.

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Help webRTC sipml5

dpkg -l | grep uuid

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