我已经搭建了一个基于Asterisk和sipml5的WebRTC系统，我可以在我的智能手机（Android）上进行音频通话，但是当我启用视频时，呼叫者可以在大约5秒内获得被呼叫者的视频，而被呼叫者根本无法获得视频. Asterisk 中是否需要任何设置？
sip.conf：
[2004]
type=friend
defaultuser=2004
username=2004
host=dynamic
secret=pass
encryption=yes
avpf=yes
icesupport=yes
context=rtc-01-dev.demo.net
directmedia=no
transport=udp,ws,wss
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass
allow=vp8,h264
nat=yes
[2005]
type=friend
defaultuser=2005
username=2005
host=dynamic
secret=pass
encryption=yes
avpf=yes
icesupport=yes
context=rtc-01-dev.demo.net
directmedia=no
transport=udp,ws,wss
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass
allow=vp8,h264
nat=yes
extensions.conf：
[rtc-01-dev.demo.net]
exten => _200Z,1,Dial(SIP/${EXTEN},30)
exten => _200Z,2,Congestion
exten => _200Z,102,Busy