asterisk-11.5.0 linphone ice

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在 https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support

看到ICE support is only used for communication between a remote endpoint and Asterisk。

终端之间的ice协商需要主被叫之间自己协商，我使用2个windows版的linphone，都打开ice，发现主叫确实发出带有ice参数的invite,可是asterisk转发invite时，把sdp中的ice参数全部去掉了，这样被叫回的200 ok里面也没有ice参数。

怎么才能让asterisk不修改invite中的sdp?????????? nat = auto_force_rport,auto_comedia icesupport = no directmedia=yes directrtpsetup=yes

终端之间的ice 协商过程是不需要voip server参与的。
asterisk11中的ice是为了asterisk作为终端时，实现p2p call。
比如asterisk注册到一个sip proxy上面，proxy的另一个exten拨打asterisk，这时实现exten和asterisk之间的p2p call。

asterisk是支持ice,但是只是支持asterisk之间，或者软电话和asterisk之间的p2p。

nat 参数好像只是处理rport，用于信令穿越nat，

Comedia mode means that Asterisk will ignore the IP and port in the received SDP from the peer and will wait for incoming RTP. This RTP should arrive to the port that Asterisk replied in the "200 OK" SDP. After that, Asterisk already knows where to send its RTP.
This make communication possible with UA's behind NAT which don't solve NAT problem in client side (STUN, ICE, ALG enabled router, etc). This options works properly in conjuntion with qualify=yes option in order to keep open the connection from Asterisk to the peer behind NAT.

directmedia 用于媒体穿越nat

directrtpsetup 让asterisk工作类似proxy，不修改sdp

asterisk位于公网 linphone位于nat后面
linphone ice=no ast ice=no directmedia=no directrtpsetup=yes 会修改sdp 中转媒体没有reinvite 有声音
linphone ice=no ast ice=no directmedia=yes directrtpsetup=yes 不会修改sdp 不中转媒体没有reinvite 没声音
linphone ice=yes ast ice=no directmedia=yes directrtpsetup=yes 会修改sdp,去掉ice 不中转媒体没有reinvite 没声音
linphone ice=yes ast ice1=yes directmedia=yes directrtpsetup=yes 会修改sdp,去掉ice 不中转媒体没有reinvite 没声音
linphone ice=yes ast ice2=yes directmedia=yes directrtpsetup=yes 会修改sdp,去掉ice 不中转媒体没有reinvite 没声音
linphone ice=yes ast ice12=yes directmedia=yes directrtpsetup=yes 会修改sdp,去掉ice 不中转媒体没有reinvite 没声音