hello all,
I am struggling with asterisk video codec.
Currently i am using sipml5 for webrtc calling in asterisk.I want to enable video call in sipml5.so which codec to be used for video calling where i want to download or purchase?
Please reply as soon as possible
Asterisk does not perform video transcoding, so there are no codec implementations for video included in the tree. For WebRTC it can pass through H.264, VP8, and VP9.
where i can download VP8 codec for asterisk 15?
There is no codec to download and install. Asterisk already knows of it using the “vp8” name. For example,
allow=!all,g722,vp8
Would allow G.722 and VP8.
As you said i can follow but the normal audio call is working in webrtc
My peer is as follows:
[102]
callerid=<102>
secret=102
context=from-internal
subscribecontext=from-internal
dial=SIP/102
mailbox=102@device
type=friend
host=dynamic
disallow=all
videosupport=yes
allow=!all,g722,vp8
dtmfmode=rfc2833
canreinvite=no
host=dynamic
trustrpid=no
sendrpid=no
directmedia=no
nat=force_rport,comedia
port=5060
qualify=yes
transport=wss,ws,udp
avpf=yes
force_avp=yes
icesupport=yes
encryption=yes
insecure=port,invite
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/public/xxx.pem
dtlsprivatekey=/etc/asterisk/public/xxx.pem
tlscertfile=/etc/asterisk/public/xxx.pem
tlsprivatekey=/etc/asterisk/public/xxx.pem
dtlssetup=actpass
rtcp_mux=yes
You’ll need to be specific in what exactly you are doing, what you are trying to do, and what is happening.
ok @jcolp i am clear in what i am doing and happening …
I am successful with the audio calling through webrtc in asterisk 15 …
Now i want to enable the video calling for that i need some guidance.the video calling is not passed through Vp8 codec…it just shows black screen and the audio flows between peers
It may be that the browser these days requires bundle and the non-bundle code is broken, in which case you’d have to use PJSIP. I use video daily with PJSIP on WebRTC including for conferences and it is fine, but chan_sip I don’t know the state of.
hi,
i am getting below error while calling from pjsip endpoints
[Mar 29 12:06:57] ERROR[14124]: res_rtp_asterisk.c:1999 ast_rtp_dtls_set_configuration: Specified private key for RTP instance ‘0x7fb08c089280’ could not be used
[Mar 29 12:06:57] ERROR[14124]: res_pjsip_sdp_rtp.c:901 setup_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance ‘0x7fb08c089280’
If you have specified a DTLS private key, then for some reason it could not be loaded and used.
I have only specified cert & ca file in my configuration as below
dtls_cert_file=/etc/asterisk/keys/certificate.crt
dtls_ca_file=/etc/asterisk/keys/ca_bundle.crt