Called with a SDP without ice-ufrag and ice-pwd

Asterisk Asterisk Support
1

Hello everybody,

I am trying to connect the SIPml5 web client to a mobile phone through asterisk 12.01, when I use xlite everything is fine both sides have audio no problem, but when i use the SIPml5 client it connects and rings the phone, but when i pick up the call, both ends have no sound (not even an echo just dead sound), i have read some of the topics on this forum sadly none of them seemed to help me, so I am asking you, if you could please help me with my problem. (OS Arch linux)
Error message from the javascript debug:

onSetRemoteDescriptionError
Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd.
tsk_utils_log_error

Similiar posts that didnt help:
=> http://forums.asterisk.org/viewtopic.php?p=198268
=> http://forums.digium.com/viewtopic.php?p=199275

My config files:
sip.conf

[general]
qualify=yes
icesupport=yes
context=default                 
allowguest=no                
allowoverlap=no                
realm=doubango.org             
udpbindaddr=0.0.0.0:5060       
tcpenable=yes 	              
tcpbindaddr=0.0.0.0             
transport=udp,tcp,ws,wss       
srvlookup=yes                   
disallow=all           
allow=ulaw

users.conf

[1060]                                          ; used to call the mobile phone
type=friend
username=1060
host=dynamic
secret=1060
context=default
hasiax=no
hassip=yes
encryption=yes
avpf=no
icesupport=yes
vieosupport=no
directmedia=no
transport=ws, udp
nat=force_rport, comedia
qualify=yes

[1061]
type=friend
username=1061
host=dynamic
secret=1061
context=default
hasiax=no
hassip=yes
encryption=yes
avpf=yes
icesupport=yes
vieosupport=no
directmedia=no
transport=ws, udp

[100]                                                    ; GSM Gateway
nat=force_rport, comedia
qualify=yes
type=friend
host=dynamic
context=default
username=100
secret=100
fromdomain=10.168.1.3
icesupport=yes
vieosupport=no

[1062]
type=friend
username=1062
host=dynamic
secret=1062
context=default
hasiax=no
hassip=yes
icesupport=yes
transport=ws, udp

extensions.conf is basically empty because our application creates them at the start of the call
example when we try to call number 123456789 to code creates this extension

exten => _123456789,1,MixMonitor(SIP/4_123456798.wav,b)
exten => _123456789,2,Dial(SIP/123456789@10.168.1.3,,r)
exten => _123456789,3,Hangup()

rtp.conf

[general]
rtpstart=10001
rtpend=20000
icesupport=true
stunaddr=stun.l.google.com:19302

Hope you can help me i have been stuck with this problem for a week :frowning:

2

A little update I looked up this issue
=> https://issues.asterisk.org/jira/browse/ASTERISK-23425?jql=text%20~%20%22ice-ufrag%22

in that issue i tried to follow the tutorial provided by Rusty Newton
=> https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

and although the asterisk CLI result was the same, I still couldn´t hear no sound… (i allowed the gsm codec in the sip.conf), i got basically the same error as my above post in the javascript console

Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd.
tsk_utils_log_error
tmedia_session_jsep01.onSetRemoteDescriptionError
(anonymous function)
3

Try with the latest 12 version, and be sure those libraries are installed before the pjsip stack.