I have integrated sipml5 with my asterisk server (which is publicly accessible).Am facing a one way audio on call between two sipxml5 clients.
Additional Information
1.Works perfectly if one client is xlite and other is sipml5
2.ICE enabled in both sipml5 clients and asterisk
3.Both clients are in different tabs of chrome
4.my sip configuration
transport=udp,ws,wss
qualify=yes
encryption = yes
avpf = yes
directmedia = outgoing
allow=ulaw
type=friend
dtmfmode=rfc2833
insecure=invite,port
qualify=yes
host=dynamic
call-limit=1
alwaysauthreject=yes
nat=force_rport,comedia
5.Here is my rtp log in asterisk
Sent RTP packet to xxx.xx.xxx.x:57214 (via ICE) (type 00, seq 048742, ts 397541448, len 4294967284)
Got RTP packet from xxx.xx.xxx.x:57214 (type 00, seq 013761, ts 397541345, len 000160)
Sent RTP packet to xxx.xx.xxx.x:57210 (type 00, seq 040424, ts 397541344, len 000164)
Got RTP packet from xxx.xx.xxx.x:57210 (type 00, seq 031460, ts 397541609, len 000160)
Sent RTP packet to xxx.xx.xxx.x:57214 (via ICE) (type 00, seq 048743, ts 397541608, len 4294967284)
Got RTP packet from xxx.xx.xxx.x:57214 (type 00, seq 013762, ts 397541505, len 000160)
Sent RTP packet to xxx.xx.xxx.x:57210 (type 00, seq 040425, ts 397541504, len 000164)
Got RTP packet from xxx.xx.xxx.x:57210 (type 00, seq 031461, ts 397541769, len 000160)