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wayne106
Brekeke Addict
Joined: 25 Jan 2008
Posts: 32
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 Posted: Fri Jun 20, 2008 3:17 pm Post subject: Packet loss through BSS |
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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.
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Mohney
Brekeke Talented
Joined: 20 Nov 2007
Posts: 79
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| Posted: Mon Jun 23, 2008 11:09 am Post subject: |
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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? |
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wayne106
Brekeke Addict
Joined: 25 Jan 2008
Posts: 32
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| Posted: Tue Jun 24, 2008 2:28 am Post subject: |
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| 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.
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wayne106
Brekeke Addict
Joined: 25 Jan 2008
Posts: 32
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| Posted: Tue Jun 24, 2008 9:59 am Post subject: |
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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, |
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Mohney
Brekeke Talented
Joined: 20 Nov 2007
Posts: 79
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| Posted: Wed Jun 25, 2008 1:36 pm Post subject: |
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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?? |
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