I have configured Asterisk 1.6.2 to support SIP videocall. turn on videosupport=yes(sip.conf) and add member such as:
[3000]
type=friend
context=default
host=dymamic
secret=123
username=cy
dtmfmode=rfc2833
canreinvite=no
disallow =all
allow=gsm
allow=alaw
allow=ulaw
allow=g722
allow=g726
allow=h261
allow=h263
allow=h263p
allow=h264
nat=no
and I install linphone in pc and htc phone g7,only can hear the vocie no video. Please give me some suggestion.Thanks
Use sip set debug on. Set a verbose logging level. Post the SIP messages containing the SDP for each side, and the trace output about the codec negotiation, here. Typically the SDP is carried on the INVITE and on the corresponding 200 OK.