using asterisk cant get video to work sip plugin

[alejo...@gmail.com]

i cant get video to work, only audio.

when i set debug lvl to 7 in janus.cfg i get this packets:

[6799653141383557] Incoming RTCP, bundling: this is audio (no video has been negotiated)

   Parsing compound packet (total of 32 bytes)

     #1 RR (201)

jitter=40.000000, fraction=0, loss=1

       RTCP PT 201, length: 32 bytes

  End of compound packet

[SIP] Fixing SSRCs (local 1571461056, peer 1853525314)

   Parsing compound packet (total of 32 bytes)

     #1 RR (201)

       RTCP PT 201, length: 32 bytes

  End of compound packet

im using linphone in order to try video, it shows a black screen instead of the video, i cant find how to properly config sip to enable video anywhere.

im more interested on only receiving video in the browser tough.

[Lorenzo Miniero]

This line seems to suggest no video was negotiated:

[6799653141383557] Incoming RTCP, bundling: this is audio (no video has been negotiated)

Check the SDP offer and answer on the WebRTC and on the SIP sides to see why this is happening.

L.

[alejo...@gmail.com]

janus sends this sdp offer:

v=0

o=- 940959236 940959236 IN IP4 192.168.0.12

s=Asterisk

t=0 0

a=group:BUNDLE audio

a=msid-semantic: WMS janus

m=audio 9 RTP/SAVPF 0 8 3 111 9 4 116 18 107 101

c=IN IP4 192.168.0.12

a=sendrecv

a=mid:audio

a=rtcp-mux

a=ice-ufrag:ujXA

a=ice-pwd:k5wt22H+PMJ4nIgisHCVEh

a=ice-options:trickle

a=fingerprint:sha-256 59:AD:8F:BE:DE:AE:17:40:A4:11:70:75:D4:AC:B2:8F:A0:FF:CE:98:A9:CE:09:64:AF:C5:E8:47:63:B9:12:9F

a=setup:actpass

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:9 G722/8000

a=rtpmap:4 G723/8000

a=rtpmap:116 G719/48000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:107 opus/48000/2

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:20

a=ssrc:1844040885 cname:janusaudio

a=ssrc:1844040885 msid:janus janusa0

a=ssrc:1844040885 mslabel:janus

a=ssrc:1844040885 label:janusa0

a=candidate:1 1 udp 2013266431 192.168.0.12 51565 typ host

a=end-of-candidates

in the asterisk -vvvvvr i see janus just sending audio and the other phone is just sending the 2 streams.

[Lorenzo Miniero]

As explained in the guidelines, please use pastebin/gist when pasting stuff, and don't add it inline.

There's no video m-line in that SDP, which means you're not asking for any video.

L.

[alejo...@gmail.com]

thanks for the response and sorry about that,

well i configured asterisk extensions to only allow g.722 and h264 and when i make the call, janus console respond with this

Incoming call from sip:00...@192.168.0.62!  siptest.js:207:12
Audio has been negotiated  siptest.js:214:13
Video has been negotiated

in my janus console i see this sdp offers:

https://gist.github.com/anonymous/5fe6e2729634da5e2c02a44b53fee706

and linphone gets this sdp in the depuration log:

https://gist.github.com/anonymous/324a50a15c6caf42760f6f103ea14504

and janus gateway admin log shows this

https://gist.github.com/anonymous/b491f25073212072838f2a6451c414c0

so its using h264 and now has the "has video" tag on true, after configuring asterisk to only allow h264, nice maybe ill get video.

then i receive this in console log:

isTrickleEnabled: undefined  janus.js:2498:3
isAudioSendEnabled: Object { audio: true, video: "true" }  janus.js:2421:3
Consent dialog should be on now  siptest.js:131:10
isAudioSendEnabled: Object { audio: true, video: "true" }  janus.js:2421:3
isVideoSendEnabled: Object { audio: true, video: "true" }  janus.js:2454:3
Default video setting is stdres 4:3  janus.js:1505:7
Adding media constraint: true

so it seems like its preparing to receive video, but when i check terminal log i receive this:

https://gist.github.com/anonymous/55e3fd666b564aa4dc07e27de2bb042a

m=video 0 RTP/SAVPF 120

c=IN IP4 0.0.0.0

a=inactive

a=rtpmap:120 VP8/90000

[2533148319552505] Video rejected by peer...

[2533148319552505] Video disabled via SDP

why theres vp8? if i disallowed vp8 from asterisk, i have 2 way audio, but no video at all. video rejected by peer, what peer? remote peer (linphone) is rejecting it or mozilla is setting vp8?

in the end i get a black screen either in linphone or portgo sip client, like theres an attempt to receive video.

well, in the end it seems like theres no way to use h264 to get video in webrtc?

the only way i got video from webrtc and asterisk is via vp8 codecs and sip.js with no webrtc gateway, but vp8 is not used with phones-

[alejo...@gmail.com]

my bad its a problem with ice trickle, forgive my utter ignorance. 

On Friday, October 13, 2017 at 2:24:19 PM UTC-3, alejo...@gmail.com wrote: