No Voice in webrtc call on natted Public asterisk server

Asterisk Asterisk Support
1

Dear All,

I have made successful two-way audio call using sip.js and asterisk on local lan.

Now, i am trying to setup webrtc audio call from public ip and asterisk being on private ip natted with public ip.

The problem is that my webrtc call is connecting but there is no audio on the both side.

(No Audio Call Flow)
WebRTC App (Sip.js based)<------------->Asterisk(192.168.79.207 Natted with 202.164.X.X)<--------------------------->SIP Phone (192.168.79.52)
(Public IP)

But when i use kapanga instead of webrtc app then we have both way audio successfully established.

(Both way audio)
Kapanga<------------->Asterisk(192.168.79.207 Natted with 202.164.X.X)<--------------------------->SIP Phone (192.168.79.52)
(Public IP)

Please download following files and dumps attached from:-

sendspace.com/file/qnsm54
or

groups.google.com/forum/#!topic … 5FZ3ApX3d8

  1. sip.conf
  2. wireshark trace for webrtc to sip call (no audio case)
  3. wireshark trace for sip to sip call (both way audio case)
  4. Kapanga Sip Log
  5. Chrome Browser Log
  6. webrtc application (call.js)

Please help me to resolve the issue.

Regards,
Raghuvendra Kumar[

2

sip.conf:-

[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port for unencrypted UDP
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
realm=202.164.x.x
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
transport=udp

;rtpkeepalive=5

localnet=192.168.79.51/255.255.255.0
localnet=192.168.79.52/255.255.255.0
localnet=192.168.79.53/255.255.255.0
localnet=192.168.79.54/255.255.255.0
localnet=192.168.79.55/255.255.255.0
localnet=192.168.79.56/255.255.255.0
localnet=192.168.79.57/255.255.255.0
localnet=192.168.79.58/255.255.255.0
localnet=192.168.79.59/255.255.255.0
localnet=192.168.79.60/255.255.255.0
localnet=192.168.79.61/255.255.255.0
localnet=192.168.79.62/255.255.255.0
localnet=192.168.79.63/255.255.255.0
localnet=192.168.79.64/255.255.255.0
localnet=192.168.79.65/255.255.255.0

canreinvite=no
externip=202.164.x.x

;media_address =202.164.x.x
;media_address =192.168.79.1

;externip=spicetalk.in
;externip=122.252.232.5

nat=yes ; Always ignore info and assume NAT
qualify=yes

;rtptimeout=15
;rtpkeepalive=15
;registerattempts=10

;stunaddr=stun.callwithus.com
[authentication]

basic-options ; a template
dtmfmode=rfc2833
context=from-office
type=friend

natted-phone ; another template inheriting basic-options
nat=yes
canreinvite=no
host=dynamic

public-phone ; another template inheriting basic-options
nat=yes
canreinvite=yes

my-codecs ; a template for my preferred codecs
disallow=all
; allow=ilbc
; allow=g729
allow=gsm
allow=g723
; allow=ulaw

ulaw-phone ; and another one for ulaw-only
disallow=all
allow=ulaw

[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
nat=force_rport,comedia
;externip=202.164.x.x

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default
directmedia=no
;nat=no
;icesupport=yes
;localnet=192.168.79.0/255.255.255.0
;externip=202.164.x.x

[1062] ; This will be the legacy SIP client
type=friend
username=1062
host=dynamic
secret=password
context=default
directmedia=no
;nat=no
icesupport=yes