BCM IP Phones Call Quality- VPN Connection

[Szewczyk]

We're getting a lot of drop outs and other problems associated with IP phones in a remote location. Is there an optimal setup for slow connections or something I could try to make the call quality go up?

We're working on the firewall now to up the packet priority to the top, but I'd like to know if there's anything I can do on the BCM400 (3.7) to improve the quality of the call?

Also, we're running into something else odd. We're trying to connect without using the VPN connection. It's easier for us to improve a single protocol's priority if it is not going through our VPN. Even with the firewall wide open, we're still not able to get past the "Re-initializing" stage on a soft phone.

Is it possible that there's a configuration setting on the BCM that I need to make to allow access from any IP on the internet? We're not using any of the Firewall or network stuff on the BCM. It's all handled by our Sonicwall firewall.

Thanks!

 

[gberger]

What codec are you using? If G711 then use G729 or G723 and use small jitter, it should help.

 

[Szewczyk]

Small jitter? Not large? I figured the larger the buffer the better. Is that wrong?

 

[SupportDude]

It kinda depends on your network.
In voice over IP (VoIP), jitter is the variation in the time between packets arriving, caused by network congestion, timing drift, or route changes.
So, if there is a lot of congestion and delay in the network, large is good.
By setting the buffer to small, you may end up with some dropped packets in a slow network, which could be your problem. Try dropping to a codec that offers more compression and uses less bandwidth as gberger suggested.

I'd suspect your connection issue is probably based on the Sonic NAT settings. You must use Full Cone NAT and hairpinning.

-SD-

http://www.mysupportguy.net

 

[gwebster]

Additional comments. Increasing jitter buffer adds to the end-to-end delay which affects perceived voice quality (think walkie-talkie and having two people talk over each other). Using compressed codecs saves bandwidth but also provides a lower perceived voice quality especially if you are having packet loss on the connection.