I am using asterisk 18.18.0 in chan_sip mode and webrtc, with a Cisco 8845 phone loaded with SIP.
When the phone calls me I get video - all good.
when I call the 8845 I only get audio.
H264 codec is specified on both.
FOR me - calling the 8845
sip debug shows:
Adding video codec h264 to SDP
Capabilities: us - (ulaw|opus|vp8|h264), peer - audio=(ulaw|opus)/video=(h264)/text=(nothing), combined - (ulaw|opus|h264)
– Executing [video@transfers-video:1] NoOp(“SIP/VC-jerry-0000002f”, “Video Call to SIP/5006 Codecs: h264,ulaw,alaw,gsm”) in new stack
Adding video codec h264 to SDP
Capabilities: us - (ulaw|alaw|gsm|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
For the 8845 calling me:
Capabilities: us - (ulaw|alaw|gsm|h264), peer - audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Adding video codec h264 to SDP
Capabilities: us - (ulaw|opus|vp8|h264), peer - audio=(ulaw|opus)/video=(h264)/text=(nothing), combined - (ulaw|opus|h264)
Adding video codec h264 to SDP
Thoughts?
Jerry