Packet loss through BSS

wayne106 · Post by **wayne106** » Fri Jun 20, 2008 3:17 pm

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

Mohney · Post by **Mohney** » Mon Jun 23, 2008 11:09 am

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?

wayne106 · Post by **wayne106** » Tue Jun 24, 2008 2:28 am

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

wayne106 · Post by **wayne106** » Tue Jun 24, 2008 9:59 am

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,

Mohney · Post by **Mohney** » Wed Jun 25, 2008 1:36 pm

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??