【发布时间】:2018-07-27 09:38:08
【问题描述】:
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,调用者可以在大约5秒内获取被调用者的视频,而被调用者无法获取视频。 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
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