Sound issues with IAX

brendon_eby
Joined: Wed 09 of Feb, 2011

Sound issues with IAX

Posted:Wed 09 of Feb, 2011 (03:02 UTC)
Ok heres the situation that has been driving me crazy for a few days:

I have asterisk 1.8 installed on a server in the amazon EC2 cloud for use in a call center I am building. I am using IAX to connect softphones to the server as we had NAT issues before when using SIP (both the server and most of the emplyees are behind a NAT network). All outbound calls go through a SIP trunk set up to Voip Innovations. I have our employees using zoiper as a softphone.

We have been having sound issues during calls, specifically 2:

First, some people have had problems with calls breaking up, being filled with static, or having one way (in) audio (as in the other party couldnt hear them, though they could hear fine). The fact that this problem only exists for some people suggests that it is a problem with their setup. However, they all have decent connection speed (the slowest is a 2mb connection being used by only 1 person), decent computers only running zoiper, skype, and a browser, the server CPU never exceeds 60 percent, the maximum number of simultaneous users is 6, and I am assuming amazon has a decent internet connection to its cloud. Furthermore, none of them have a problem when calling through skype. Any ideas? The only thing I can figure out is that them all being in colombia might cause a problem.

My other problem is only had by 1 person, but if someone is having this issue now, others might in the future. His outbound audio is very quiet, and sounds echoy as though he is on speaker phone. Most people cannot hear him very well (not silent though). Again, it works perfectly through skype on the same setup at the same time. I also had him try using the same headset, account, and softphone on a different computer (different network as well), and it worked fine, suggesting something wrong with his computer/network. I have tried playing with different codecs and gains without any change. At first I thought it might be something about transferring an IAX user to a SIP trunk, but the same problem exists on internal calls. SIP doesnt work for him at all, im assuming because of NAT issues. Anyone have a similar issue?

Here are my conf file settings. All users are set using the template (emp) below in iax.conf. The outbound trunk is set in sip.conf:

iax.conf:

[general]
autokill=yes
trunk=no

[emp](!)
type=friend
context=phones
host=dynamic
canreinvite=no


sip.conf

[general]
context=incoming
;tos=0x18
port=5060
externip=204.XXX.XXX.XX
localnet=10.0.0.0/255.0.0.0
nat=yes
qualify=yes
disallow=all
allow=gsm
;allow=g729
allow=ulaw
allow=alaw

[voip_innovations]
type=peer
host=XX.XXX.XXX.XX
context=incoming
dtmfmode=rfc2833
disallow=all
allow=gsm,ulaw
;deny=0.0.0.0/0
;permit=XX.XXX.XXX.XX/32
insecure=invite
canreinvite=no


extensions.conf:

[globals]
static=yes
writeprotect=no
clearglobalvars=no
OUTBOUNDTRUNK=Sip/voip_innovations
CALLFILENAME="outgoing-"
OUTCALLERID=<xxxxxxxxxx>

[general]
autofallthrough=yes

[outgoing]
exten => _NXXNXXXXXX,1,answer()
exten => _NXXNXXXXXX,n,Set(CALLERID(all)=${OUTCALLERID})
exten => _NXXNXXXXXX,n,Set(VOLUME(TX)=1)
exten => _NXXNXXXXXX,n,Monitor(wav,${CALLFILENAME}+${EXTEN},m)
exten => _NXXNXXXXXX,n,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten => _NXXNXXXXXX,n,hangup()



Thanks in advance!