webrtc - webrtc2sip + sipml5 + asterisk no audio issue -

webrtc - webrtc2sip + sipml5 + asterisk no audio issue -

i having problem in having sipml5 phone call other sipml5 via webrtc2sip , asterisk.

i have installed configured asterisk(version 11.10.0) + webrtc2sip(latest) + sipml5(chrome version 30.0.1599.66) phone call 1 box other on websocket.

i can create sip phone call through , reply other side seems there no audio/voice packets gets exchanged evident rtp , sip debug log , tcpdump.

asterisk , users on separate servers , found sip phone call it's sound packet not appear.

i've created 2 users(1060 , 1061) , when create phone call these asterisk response.

rtp set debug on rtp debugging enabled *cli> == spawn extension (default, 1060, 1) exited non-zero on 'sip/1061-00000000' == using sip rtp cos mark 5 -- executing [1060@default:1] dial("sip/1061-00000002", "sip/1060") in new stack == using sip rtp cos mark 5 -- called sip/1060 -- sip/1060-00000003 ringing -- sip/1060-00000003 answered sip/1061-00000002

asterisk settings are..

sip.conf

[general] port=5060 bindaddr=0.0.0.0 context=default transport=ws,wss,udp srvlookup=yes

users.conf

[1060] type=peer username=1060 host=dynamic secret=1234 context=default disallow=all allow=ulaw transport=udp,ws,wss encryption=yes avpf=yes icesupport=yes nat=yes,force_rport [1061] type=peer username=1061 host=dynamic secret=1234 context=default encryption=yes avpf=yes icesupport=yes nat=yes disallow=all allow=ulaw transport=udp,ws,wss

extensions.conf

[general] static=yes writeprotect=no [default] exten=>1060,1,dial(sip/1060) exten=>1061,1,dial(sip/1061)

rtp.conf

[general] icesupport=yes stunaddr=stun.l.google.com:19302 strictrtp=no rtcpinterval=6000 rtpchecksums=no

i can hear dialling sound on 1 end , ringing on other end phone call connected, can't hear anything.

could kindly help please.. very desperate.. in advance!

remove avpf option, not needed when using webrtc2sip. if still no voice - check sipml5 log (firefox or chrome debug console ) while sip registration , while call.

asterisk webrtc