Added video codecs to pjsip.conf:
allow=h263p
allow=h263
allow=h264
allow=vp8
see full PJSIP configuration on GitHub.
core show channeltypes
Type Description Devicestate Presencestate Indications Transfer
------------- ------------- ------------- ------------- ------------- -------------
Recorder Bridge Media Recording Channel Driver no no yes no
Announcer Bridge Media Announcing Channel Driver no no yes no
USTM UNISTIM Channel Driver no no yes no
CBAnn Conference Bridge Announcing Channel no no yes no
CBRec Conference Bridge Recording Channel no no no no
PJSIP PJSIP Channel Driver yes no yes yes
AudioSocket AudioSocket Channel Driver no no no no
UnicastRTP Unicast RTP Media Channel Driver no no no no
MulticastRTP Multicast RTP Paging Channel Driver no no no no
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes no yes yes
Local Local Proxy Channel Driver yes no yes no
Surrogate Surrogate channel used to pull channel f no no no no
----------
12 channel drivers registered.
core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME FORMAT QUALITY DESCRIPTION
------------------------------------------------------------------------------------------------
30 image png png 0 (PNG Image)
6 audio g726 g726 85 (G.726 RFC3551)
4 audio alaw alaw 100 (G.711 a-law)
2 audio g723 g723 20 (G.723.1)
20 audio speex speex 40 (SpeeX)
21 audio speex speex16 40 (SpeeX 16khz)
22 audio speex speex32 40 (SpeeX 32khz)
24 audio g722 g722 110 (G722)
25 audio siren7 siren7 85 (ITU G.722.1 (Siren7, licensed from Polycom))
31 video h261 h261 0 (H.261 video)
32 video h263 h263 0 (H.263 video)
8 audio adpcm adpcm 80 (Dialogic ADPCM)
35 video h265 h265 0 (H.265 video)
43 audio silk silk8 0 (SILK Codec (8 KHz))
44 audio silk silk12 0 (SILK Codec (12 KHz))
45 audio silk silk16 0 (SILK Codec (16 KHz))
46 audio silk silk24 0 (SILK Codec (24 KHz))
27 audio g719 g719 95 (ITU G.719)
33 video h263p h263p 0 (H.263+ video)
34 video h264 h264 0 (H.264 video)
19 audio g729 g729 20 (G.729A)
9 audio slin slin 115 (16 bit Signed Linear PCM)
10 audio slin slin12 116 (16 bit Signed Linear PCM (12kHz))
11 audio slin slin16 117 (16 bit Signed Linear PCM (16kHz))
12 audio slin slin24 118 (16 bit Signed Linear PCM (24kHz))
13 audio slin slin32 119 (16 bit Signed Linear PCM (32kHz))
14 audio slin slin44 120 (16 bit Signed Linear PCM (44kHz))
15 audio slin slin48 121 (16 bit Signed Linear PCM (48kHz))
16 audio slin slin96 122 (16 bit Signed Linear PCM (96kHz))
17 audio slin slin192 123 (16 bit Signed Linear PCM (192kHz))
3 audio ulaw ulaw 100 (G.711 u-law)
18 audio lpc10 lpc10 25 (LPC10)
42 audio none none 0 (<Null> codec)
41 image t38 t38 0 (T.38 UDPTL Fax)
38 video vp9 vp9 0 (VP9 video)
37 video vp8 vp8 0 (VP8 video)
5 audio gsm gsm 60 (GSM)
36 video mpeg4 mpeg4 0 (MPEG4 video)
23 audio ilbc ilbc 45 (iLBC)
39 text red red 0 (T.140 Realtime Text with redundancy)
40 text t140 t140 0 (Passthrough T.140 Realtime Text)
28 audio opus opus 50 (Opus Codec)
29 image jpeg jpeg 0 (JPEG image)
7 audio g726aal2 g726aal2 85 (G.726 AAL2)
1 audio codec2 codec2 0 (Codec 2)
26 audio siren14 siren14 90 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
Codecs in Linphone desktop client:
mobile Linphone client:
Tried to call from mobile client (C) do desktop client (A) and enable video, but the video was not displayed on client A, see the logs:
I was able to find video and H264 in the logs:
v=0
o=neo 1745 1016 IN IP4 13.163.232.112
s=Talk
c=IN IP4 13.163.232.112
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 58875 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp:42841
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
m=video 48271 RTP/AVP 96 97
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtpmap:97 VP8/90000
a=fmtp:97
a=rtcp:56434
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 ccm fir
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 nack sli
a=rtcp-fb:97 ack rpsi
a=rtcp-fb:97 ccm fir
A call from C to another desktop client with VP8 codec:
v=0
o=neo 2172 2315 IN IP4 13.163.232.112
s=Talk
c=IN IP4 13.163.232.112
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 51280 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp:45829
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
m=video 48939 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=fmtp:96
a=rtcp:59925
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 nack sli
a=rtcp-fb:96 ack rpsi
a=rtcp-fb:96 ccm fir
Capabilities does not contain video codecs. What does it mean?
core show channeltype PJSIP
-- Info about channel driver: PJSIP --
Device State: yes
Presence State: no
Indication: yes
Transfer : yes
Capabilities: (codec2|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin12|slin16|slin24|slin32|slin44|slin48|slin96|slin192|lpc10|g729|speex|speex16|speex32|ilbc|g722|siren7|siren14|g719|opus|silk8|silk12|silk16|silk24)
Digit Begin: yes
Digit End: yes
Send HTML : no
Image Support: no
Text Support: yes
No video in Asterisk w/ PJSIP
https://community.asterisk.org/t/no-video-in-asterisk-w-pjsip/67018
However, it appears that PJSIP is not capable of sending any kind of video?
How to enable video in Asterisk 22
https://community.asterisk.org/t/how-to-enable-video-in-asterisk-22/106972/1
PJSIP video configuration
https://community.asterisk.org/t/pjsip-video-configuration/82415
There is no “videosupport” option, it has to be removed or the endpoint won’t be loaded. That option is for chan_sip only. You only need to specify a video codec in chan_pjsip.
You can set multiple however Asterisk does not transcode video so both sides have to use the same one.