Sound Quality (GSM, IAX2, Teliax)

[djn602]

Hello,

I am pretty new to VoIP... I finally got my system up and running and have been placing many phone calls. The people on the other end are complaining of some pauses and pops before they hear me and some dropouts. I have been tinkering with settings to see if things improve but to no avail. I am currently setting my box up to do SIP instead. Any suggestions as to what to tinker with to help this issue? Teliax says aterisk 1.2.1 to 1.2.4 has "audio" issues. Has any one else heard of this? They recommend going back to 1.2.0-rc2.

My IAX2 config:

allow=gsm
auth=md5
context=from-pstn
disallow=all
forcejitterbuffer=no
host=voip-co2.teliax.com
jitterbuffer=yes
maxjitterbuffer=800
maxjitterinterps=20
nat=yes
qualify=1000
resyncthreshold=1000
secret=xxxxxxx
tos=lowdelay
type=friend
username=xxx

Thanks,
Dave

[datarax]

download pingplotter from www.pingplotter.com and trace your connection to them. Let it run for a few hours to graph the connection. do you have an ata or sip phone? Register it to them and make calls and see how it performs. That will tell you if it is * or not. Where are you located?

[djn602]

I have ping plotter running now. So far it is 45ms. I am in Springfield Missouri on SBC Adsl with 384-512kbps upstream. The hope is to get all working well and use this for the company I work for, around 8 sim calls max (right now). Also I have a Grandstream GXP-2000 for testing so I could use it to bypass *, didn't think of that.

Also I put someone on hold for a bit and the music did not suffer the same issues, weird.

Thanks

[datarax]

If you have a good connection then look at the codecs. Try ulaw uncompressed first then work your way down the compression codecs. Ulaw is the least taxing on the * box but the most bandwidth intensive. It looks like you should be able to support 3 calls with ulaw so 1 should give you no trouble.

[mberlant]

As you discovered, ping times are generally unrelated to voice quality. The time it takes for a 32 byte packet to get between you and the other end bears no relation to how thousands of RTP packets transmitted back to back will fare. You may have 384kbps (good for 3-4 G.711 conversations) with SBC, but there is no guarantee as to what is happening along the route between SBC and TelIAX.

I would recommend experimenting with different CODECs to find the one that balances acceptable call quality with efficient use of bandwidth. Start by trying out G.711 just to eliminate the possibility that there is some hardware/software/system problem mucking up your voice quality in general.