[asterisk-users] out of the blue one way audio

[Tarek Sawah]

Greetings List.
we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following.

1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server.
2- Internet link of 1Mbps Dedicated Leased Line.
3- Cisco Router
4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel(R) Xeon(R) X3210  @ 2.13GHz CPU)
5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip).
6- Packet8 Sip trunking for Inbound calls
7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs)

Network Profile:
Cisco Router has a Public IP of 196.XXX.XXX.XXX  and a private IP 192.168.100.245
computers have IP addresses : 192.168.100.XXX/24
default gateway: 192.168.100.245
DC: 192.168.100.2
DNS: 192.168.100.2
PROXY Server: 192.168.100.2  (Forced in Internet Explorer)
Voip Traffic going directly from 192.168.100.245
Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection)

Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through.


Asterisk Side:
sip.conf sample:
[GENERAL]
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
t38pt_udptl = yes
bindport=5070
externip=SERVER_IP
rtptimeout=60
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas
rtpholdtimeout=120
rtpkeepalive=20
allow=gsm
t38pt_udptl=yes
sendrpid=yes
trustrpid=no
directrtpsetup=yes

[USERNAME]
deny=0.0.0.0/0.0.0.0
type=friend
secret=PASSWORD
qualify=yes
port=5060
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=gsm
context=from-callcenter
canreinvite=no


we have a call recording for outbound and inbound calls.
the problem is not happening on all calls at once.. it happens on random
 extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) 

one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register
we changed the port to 5070 and things are working properly now.
although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060.

any additional information can be provided.

 
Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993