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[asterisk-users] PJSIP - Video Support for WebRTC

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odongfack at yahoo.fr
Guest

Posted: Tue Jul 26, 2016 8:30 am Post subject: [asterisk-users] PJSIP - Video Support for WebRTC

Matthew Jordan <mjordan <at> digium.com> writes:

Quote:

On Mon, Mar 23, 2015 at 8:55 AM, Gosmac <goseeped <at> gmail.com>

wrote:

Quote:

Hey i have an interesting topic to discuss here.

The main goal here is to be able to make a video call between two

WebRTC endpoints registered on asterisk 13

Quote:

it is a feature that definitely asterisk 13 should support .

Quote:

the problems that i faced with this is the following and i hope i

could get an advise here.

Quote:

asterisk 13 vanilla version has some issues marking the video

packets this complain web browser

Quote:

specially VP8 codecs so a friend of mine help me to patch

res_rtp_asterisk and now asterisk is marking

Quote:

video streams Smile

it just mark video packets not touch anything else

and web browser show video on web page

Quote:

now I’m using online demo http://tryit.jssip.net/ is stable and get

more updates than sipml5. so i try

Quote:

echo() dialplan test and everything work perfect on echo test Smile

Quote:

i have two questions and i hope you could give me some advise.

1) after marking video packet I’m able to make Dial() between two

webrtc peers but i get one way audio and

Quote:

video on callee party, “after 3 minutes on call” i get two way audio

and video on all parties seems to be

Quote:

not just a problem on a missing keyframe.

Quote:

1.1) the 3 minutes delay only happen using chrome stable , could be

a dtls problem when asterisk make an

Quote:

offer to other endpoint?

Quote:

1.2) when i use chrome-dev and i disable dlts encryption everything

work perfect on video call.

Quote:

2) after marking video packets i realize that when you make a call

with video and you involve on dialplan an

Quote:

application like playback or music on hold any application that

played audio files (audio and video never work).

Quote:

2.1) asterisk is muggling the audio and video streams ?

This is good information for all guys out there that wants to

support video on webrtc in asterisk 13

Quote:

Please stop spamming the list with this e-mail. Resending it multiple
times is clearly not yielding the results you'd like.

Hi Matthew,
I'm testing WebRTC (JSSIP) with Asterisk 12.8 after following the
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support link.
Using Firefox, I can connect both JSSIP Clients to asterisk. When I Call
one Client, the Client just Ring One Time and after pick up a receive
WebRTC error on the Firefox browser.
Here is my asterisk sip debug:

<--- SIP read from WS:192.168.2.103:49851 ---> INVITE
sip:6000@192.168.2.106 SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
Max-Forwards: 69
To: <sip:6000@192.168.2.106>
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6407 INVITE
X-Can-Renegotiate: false
Contact: <sip:5krseuop@0iemcrsq9tm0.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 3158

v=0
o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
C1:23:72:03:F6:61:CC:F6
a=group:BUNDLE sdparta_0 sdparta_1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56806 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56807 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
56810 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
56811 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
192.168.2.103 rport 56808
a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
192.168.2.103 rport 56812
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_0
a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
{bba6da45-42c8-4529-8f4b-046cffcdc40d}
a=rtcp:56812 IN IP4 87.169.189.102
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56814 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56815 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
60291 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
60292 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 64504 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 64505 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56816 typ srflx raddr
192.168.2.103 rport 56816
a=candidate:5 2 UDP 1685856254 87.169.189.102 64504 typ srflx raddr
192.168.2.103 rport 64504
a=recvonly
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:126
profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
a=ice-pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_1
a=rtcp:64504 IN IP4 87.169.189.102
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-mux
a=rtpmap:120 VP8/90000
a=rtpmap:126 H264/90000
a=rtpmap:97 H264/90000
a=setup:actpass
a=ssrc:3124982 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
<------------->
--- (14 headers 71 lines) ---
Using INVITE request as basis request - 1ansppdrpdulbtr3j5ub Found peer
'6001' for '6001' from 192.168.2.103:49851

<--- Reliably Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 401
Unauthorized
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK8394689;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as28bc71f3
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6407 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.2.106",
nonce="69dcc467"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '1ansppdrpdulbtr3j5ub' in 32000 ms
(Method: INVITE)

<--- SIP read from WS:192.168.2.103:49851 ---> ACK
sip:6000@192.168.2.106 SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
To: <sip:6000@192.168.2.106>;tag=as28bc71f3
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6407 ACK
Content-Length: 0

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

<--- SIP read from WS:192.168.2.103:49851 ---> INVITE
sip:6000@192.168.2.106 SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783
Max-Forwards: 69
To: <sip:6000@192.168.2.106>
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 INVITE
Authorization: Digest algorithm=MD5, username="6001",
realm="192.168.2.106", nonce="69dcc467", uri="sip:6000@192.168.2.106",
response="845417814e71ce56e93d846538ea31ae"
X-Can-Renegotiate: false
Contact: <sip:5krseuop@0iemcrsq9tm0.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 3158

v=0
o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
C1:23:72:03:F6:61:CC:F6
a=group:BUNDLE sdparta_0 sdparta_1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56806 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56807 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
56810 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
56811 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
192.168.2.103 rport 56808
a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
192.168.2.103 rport 56812
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_0
a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
{bba6da45-42c8-4529-8f4b-046cffcdc40d}
a=rtcp:56812 IN IP4 87.169.189.102
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102
a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
56814 typ host
a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
56815 typ host
a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0
2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
60291 typ host
a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
60292 typ host
a=candidate:4 2 UDP 2122055934 192.168.2.103 64504 typ host
a=candidate:6 2 UDP 2122252542 192.168.56.1 64505 typ host
a=candidate:5 1 UDP 1685856255 87.169.189.102 56816 typ srflx raddr
192.168.2.103 rport 56816
a=candidate:5 2 UDP 1685856254 87.169.189.102 64504 typ srflx raddr
192.168.2.103 rport 64504
a=recvonly
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:126
profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
a=ice-pwd:138f583004cb3079134e8e8f20dac36f
a=ice-ufrag:0941ac54
a=mid:sdparta_1
a=rtcp:64504 IN IP4 87.169.189.102
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-mux
a=rtpmap:120 VP8/90000
a=rtpmap:126 H264/90000
a=rtpmap:97 H264/90000
a=setup:actpass
a=ssrc:3124982 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
<------------->
--- (15 headers 71 lines) ---
Using INVITE request as basis request - 1ansppdrpdulbtr3j5ub Found peer
'6001' for '6001' from 192.168.2.103:49851
== TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
enabled
== TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
enabled
== Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found audio description format opus for ID 109 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 RTP video format 120 Found
RTP video format 126 Found RTP video format 97 Found video description
format VP8 for ID 120 Found video description format H264 for ID 126
Found video description format H264 for ID 97
Capabilities: us -
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aa
l2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren
14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin19
2|opus|vp8|silk8|silk12|silk16|silk24),
peer - audio=(ulaw|alaw|g722|opus)/video=(h264|vp8)/text=(nothing),
combined - (ulaw|alaw|g722|h264|opus|vp8) Non-codec capabilities (dtmf):
us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0
(nothing) Peer audio RTP is at port 87.169.189.102:56808 Peer doesn't
provide T.140 Looking for 6000 in outgoing (domain 192.168.2.106)
list_route: route/path hop:
<sip:5krseuop@0iemcrsq9tm0.invalid;transport=ws;ob>

<--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 100 Trying
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@192.168.2.106:5060;transport=WS>
Content-Length: 0

<------------>
-- Executing [6000@outgoing:1] Dial("SIP/6001-00000000",
"SIP/6000") in new stack
== TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
enabled
== TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
enabled
== Using SIP RTP CoS mark 5
We think we can do text
And we have a text rtp object
Audio is at 17504
Lets set up the text sdp
Text is at 0.0.0.0:12836
Adding codec 100003 (ulaw) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100021 (slin16) to SDP
Adding text codec 400001 (red) to SDP
Adding text codec 400002 (t140) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100028 (speex32) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100030 (opus) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting
(NAT) to 192.168.2.103:49848:
INVITE sip:bbglnljp@72rvpk435t95.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
Max-Forwards: 70
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>
Contact: <sip:6001@192.168.2.106:5060;transport=WS>
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.8.2
Date: Sun, 24 Jul 2016 22:21:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 2778

v=0
o=root 1723742189 1723742189 IN IP4 192.168.2.106 s=Asterisk PBX 12.8.2
c=IN IP4 192.168.2.106 t=0 0 m=audio 17504 RTP/SAVPF 0 4 3 8 111 5 7 18
110 117 97 112 9 118 102 115
116 119 107 96 108 109 113 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:119 speex/32000
a=rtpmap:107 opus/48000/2
a=fmtp:107
maxplaybackrate=48000;sprop-
maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-
stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:108 SILK/12000
a=fmtp:108 maxaveragebitrate=12000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:109 SILK/16000
a=fmtp:109 maxaveragebitrate=20000
a=fmtp:109 usedtx=0
a=fmtp:109 useinbandfec=1
a=rtpmap:113 SILK/24000
a=fmtp:113 maxaveragebitrate=30000
a=fmtp:113 usedtx=0
a=fmtp:113 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=ice-ufrag:29c82286064fb054206fb5686f954dde
a=ice-pwd:668a84152d94daf26a719ce95a2174fe
a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 17504 typ host
a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 17504 typ srflx
raddr 192.168.2.106 rport 17504 a=candidate:Hc0a8026a 2 UDP 2130706430
192.168.2.106 17505 typ host
a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 17505 typ srflx
raddr 192.168.2.106 rport 17505 a=connection:new a=setup:actpass
a=fingerprint:SHA-256
C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
86:F8:7B:1A:8D:DE:B3:47
a=sendrecv
m=text 12836 RTP/SAVPF 105 106
a=ice-ufrag:1d890d34098281a73af392833cdf7626
a=ice-pwd:06e51e922264e51c5cc0209e3a1ec6a7
a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 12836 typ host
a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 12836 typ srflx
raddr 192.168.2.106 rport 12836 a=candidate:Hc0a8026a 2 UDP 2130706430
192.168.2.106 12837 typ host
a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 12837 typ srflx
raddr 192.168.2.106 rport 12837 a=connection:new a=setup:actpass
a=fingerprint:SHA-256
C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
86:F8:7B:1A:8D:DE:B3:47
a=rtpmap:105 RED/1000
a=fmtp:105 106/106/106
a=rtpmap:106 T140/1000
a=sendrecv

---
-- Called SIP/6000

<--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 INVITE
Supported: ice,replaces,outbound
Content-Length: 0

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

<--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 INVITE
Contact: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>
Supported: ice,replaces,outbound
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: route/path hop:
<sip:bbglnljp@72rvpk435t95.invalid;transport=ws>
-- SIP/6000-00000001 is ringing

<--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 180 Ringing
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@192.168.2.106:5060;transport=WS>
Content-Length: 0

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

<--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 488 Not
Acceptable Here
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 INVITE
Supported: ice,replaces,outbound
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.2.103:49848:
ACK sip:bbglnljp@72rvpk435t95.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
Max-Forwards: 70
From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59
To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
Contact: <sip:6001@192.168.2.106:5060;transport=WS>
Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.8.2
Content-Length: 0

---
Scheduling destruction of SIP dialog
'5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060' in 32000 ms
(Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [6000@outgoing:2] Answer("SIP/6001-00000000", "") in
new stack Audio is at 19538 Adding codec 100003 (ulaw) to SDP Adding
codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding
codec 100030 (opus) to SDP

<--- Reliably Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200
OK
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@192.168.2.106:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 1055

v=0
o=root 1700582523 1700582523 IN IP4 192.168.2.106 s=Asterisk PBX 12.8.2
c=IN IP4 192.168.2.106 t=0 0 m=audio 19538 RTP/SAVPF 0 8 9 109
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:109 opus/48000/2
a=fmtp:109
maxplaybackrate=48000;sprop-
maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-
stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=ptime:20
a=maxptime:60
a=ice-ufrag:7fbfa28012692271620bb8c22da32ff3
a=ice-pwd:30067a57115528082e8744df31454da4
a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 19538 typ host
a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 19538 typ srflx
raddr 192.168.2.106 rport 19538 a=candidate:Hc0a8026a 2 UDP 2130706430
192.168.2.106 19539 typ host
a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 19539 typ srflx
raddr 192.168.2.106 rport 19539 a=connection:new a=setup:active
a=fingerprint:SHA-256
C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
86:F8:7B:1A:8D:DE:B3:47
a=sendrecv
m=video 0 UDP/TLS/RTP/SAVPF 120 126 97

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

<--- SIP read from WS:192.168.2.103:49851 ---> ACK
sip:6000@192.168.2.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK690056
Max-Forwards: 69
To: <sip:6000@192.168.2.106>;tag=as1792125e
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6408 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

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

<--- SIP read from WS:192.168.2.103:49851 ---> BYE
sip:6000@192.168.2.106:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK1426296
Max-Forwards: 69
To: <sip:6000@192.168.2.106>;tag=as1792125e
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6409 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog '1ansppdrpdulbtr3j5ub' in 32000 ms
(Method: BYE)

<--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200 OK
Via: SIP/2.0/WS
0iemcrsq9tm0.invalid;branch=z9hG4bK1426296;received=192.168.2.103;rport=
49851
From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6409 BYE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May you please help me to make it word? i'm just interessting for the
audio.
Thank you in advance
--
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simon.hohberg at mcs-d...
Guest

Posted: Wed Jul 27, 2016 7:35 am Post subject: [asterisk-users] PJSIP - Video Support for WebRTC

On 07/26/2016 03:15 PM, Olivier wrote:

Quote:

Matthew Jordan <mjordan <at> digium.com> writes:

Quote:

On Mon, Mar 23, 2015 at 8:55 AM, Gosmac <goseeped <at> gmail.com>

wrote:

Quote:

Hey i have an interesting topic to discuss here.

The main goal here is to be able to make a video call between two

WebRTC endpoints registered on asterisk 13

Quote:

it is a feature that definitely asterisk 13 should support .

Quote:

the problems that i faced with this is the following and i hope i

could get an advise here.

Quote:

asterisk 13 vanilla version has some issues marking the video

packets this complain web browser

Quote:

specially VP8 codecs so a friend of mine help me to patch

res_rtp_asterisk and now asterisk is marking

Quote:

video streams Smile

it just mark video packets not touch anything else

and web browser show video on web page

Quote:

now I’m using online demo http://tryit.jssip.net/ is stable and get

more updates than sipml5. so i try

Quote:

echo() dialplan test and everything work perfect on echo test Smile

Quote:

i have two questions and i hope you could give me some advise.

1) after marking video packet I’m able to make Dial() between two

webrtc peers but i get one way audio and

Quote:

video on callee party, “after 3 minutes on call” i get two way audio

and video on all parties seems to be

Quote:

not just a problem on a missing keyframe.

Quote:

1.1) the 3 minutes delay only happen using chrome stable , could be

a dtls problem when asterisk make an

Quote:

offer to other endpoint?

Quote:

1.2) when i use chrome-dev and i disable dlts encryption everything

work perfect on video call.

Quote:

2) after marking video packets i realize that when you make a call

with video and you involve on dialplan an

Quote:

application like playback or music on hold any application that

played audio files (audio and video never work).

Quote:

2.1) asterisk is muggling the audio and video streams ?

This is good information for all guys out there that wants to

support video on webrtc in asterisk 13

Quote:

Please stop spamming the list with this e-mail. Resending it multiple
times is clearly not yielding the results you'd like.

Hi Olivier,

I am not sure what is the issue with your current setup, however I would
like to point you to the latest Asterisk releases, i.e. >13.9.1. I think
those releases include some fixes for issues related to WebRTC. For me
theses releases work very well with WebRTC (audio & video).

There is only one remark I have to make: When using PJSIP I experienced
an issue with the maximum size for SIP messages. I therefore had to
manually increase it to be able to get it working.

Hope it helps,

Simon

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