Packet loss through BSS

1. Brekeke Product Name and version: Brekeke SIP Server , Version 2.1.6.6 Standard

2. Java version: 1 .5 .6

3. OS type and the version: Win2003 svr web edition

4. UA (phone), gateway or other hardware/software involved:
Sipura 941
I am trying to migrate from Asterisk to BSS

My BSS server is connected directly to the net with a real IP using the win2k3 firewall for protection. I have a cisco on the same subnet also directly on the net, which I using to terminate calls on the PSTN.
When I call from my Sipura (or any UA) the audio quality is not good,(useable but a bit choppy) and I see there is packet loss.

I have an asterisk server connected next to the BSS also sending calls to the cisco.
When I make calls from the same UA through asterisk (or even to asterisk, say voicemail) the audio quality is far better and no packet loss (not even 1 packet in 5mins)

I think I have ruled out network problems, as quite intensive ping test reveal no problems, I have no idea where else to turn, even if I take the cisco out of the loop and say call asterisk voicemail through BSS I get the same loss, its about 50 packets / min. I tried turning off the windows firewall to see if that would help, but it made no difference.

Anyone got any ideas? anything much appreciated, I have now completed by BSS project to replace asterisk, just niggling issues like this one holding me up.

Thanks

Are you using Brekeke PBX too in the environment?
How about the CPU usage in the Win2003 server while you are on a phone call?
Can you disable the RTP-realy at the SIP Server?

Mohney wrote:Are you using Brekeke PBX too in the environment?
How about the CPU usage in the Win2003 server while you are on a phone call?
Can you disable the RTP-realy at the SIP Server?

Hi, Thanks for the reply,
The CPU usage is nothing, its a fast box quad Xeon and so far only has 1 session in progress at a time (me testing)

Ping test result in no loss (even for 5000 pings)
I'm not using PBX only BSS, I'm going to try a phone outside of the NAT and turn RTP relay off see what happens.

W

I tried a phone that was on a real IP and disabled RTP relay and there is no loss, so its definitly BSS thats dropping the packets while its relaying. I have rebooted it, and stopped the firewall and its still the same. I might try and install a demo version on another PC and see if it does the same.

Thanks,

How about voice delay while the server is relaying RTP?

If you make a call between clients via the Brekeke SIP Server without Asterisk.., how about the audio quality??