Strange WebRTC issue with Asterisk

I have two servers, both without a NAT, one has Debian (Asterisk 11.5.0) and works fine but the second with Ubuntu Server 12.04 (Asterisk 11.6-cert) the SIPML5 doesn’t receive audio, but cannot find the difference between them two:

I read this topic that it’s similar to my issue:
viewtopic.php?f=1&t=88761
I have installed the uuid-dev and all that without success :frowning:

And followed all the instructions, but still cannot get the audio in the SIPML5

There is message in the SIP dialog in the Chrome (32 Stable):
tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.

The only part of the SIP dialog that differs is this:

v=0 o=root 1977488278 1977488278 IN IP4 188.123.231.112 s=Asterisk PBX 11.6-cert1 c=IN IP4 188.123.231.112 t=0 0 m=audio 10748 RTP/SAVPF 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:OHJ8T2u7vVnSHVzyp0Sd

My sip.conf is:

[general]
allowguest=no
udpbindaddr=0.0.0.0:5060
transport=udp,ws,wss

[extensions](!)
host=dynamic
nat=force_rport,comedia
qualify=yes
type=friend
context=extensions

[10](extensions)
secret=secretcode
disallow=all
allow=opus
transport=udp,wss,ws
encryption=yes
callerid=WebRTC <10>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no
hasiax=no
hassip=yes
videosupport=no

And I have all the config related to STUN and ICE in RTP.conf:

[general]
rtpstart=10000
rtpend=20000

icesupport=true
stunaddr=stun.l.google.com:19302

turnaddr=numb.viagenie.ca
turnusername=numb-email
turnpassword=numn-pass

Cant figure out what might be happening :frowning: Even RTP Packets are being forwarded to the correct destination:

Sent RTP packet to 95.63.247.14:62449 (type 111, seq 064972, ts 023136, len 000068) Sent RTP packet to 95.63.247.14:62449 (type 111, seq 064973, ts 024096, len 000055) Sent RTP packet to 95.63.247.14:62449 (type 111, seq 064974, ts 025056, len 000056) Sent RTP packet to 95.63.247.14:62449 (type 111, seq 064975, ts 026016, len 000055)

[quote]There is message in the SIP dialog in the Chrome (32 Stable):
tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. [/quote]Your asterisk is not compiled with ICE support enable the uuid or libuuid and the dev packages, re run the configure script and recompile it.

Hello navaismo

I did reconfigure/recompile with libuuid already installed:

i libuuid1 - Universally Unique ID library

and also uuid-dev

i uuid-dev - universally unique id library - headers and static libraries

I configure Asterisk with this parameters:

./configure --with-crypto --with-ssl --with-srtp

And in the menuselect I cant find anything related directly with ICE so I think is being compiled with ICE correctly.

If the server isn’t sending then there is no ice, try installing all dependencies with the install_prereq script, also enable ice in sip peer and rtp.conf

Thanks for the support navaismo

But after long hours of testing I’ve found the problem appears when I compile Asterisk with OPUS support applying this patch:

github.com/meetecho/asterisk-opus

In Asterisk 11.5 it works fine, but Asterisk >11.6 it provokes an error during the Audio Call (RTP communication and ICE issue)