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Unread 04-03-2009, 06:04 PM   #1
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Default Voice is breaking today

Hi all,

I do not have problems in the last days, but today, when i call from pstn to my sip number, the voice pstn-->sip it's ok, but the voice sip--->pstn have breakings.

¿friday and my isp is satured?

anyone haves ideas to improve the quality.

regards
f_
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Unread 04-04-2009, 08:37 AM   #2
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Sometimes it helps to use a low bit rate codec such as G729a. What kind of IP phone are you using? Do you know if RTP packets in the sip--->pstn direction are coming direct from the source, or through Voxalot?
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Unread 04-04-2009, 03:48 PM   #3
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Quote:
Originally Posted by boatman View Post
Sometimes it helps to use a low bit rate codec such as G729a.
Where i can see what codec i'm using?

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What kind of IP phone are you using?
LINK1 LINK2

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Do you know if RTP packets in the sip--->pstn direction are coming direct from the source, or through Voxalot?
I don't know.
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Unread 04-04-2009, 06:46 PM   #4
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Quote:
Originally Posted by 137383 View Post
Where i can see what codec i'm using?
Look at the codec settings in your ATA configuration. To be sure of using just one codec (for testing purposes), disable the others.

The audio packets may flow better if they are not coming through Voxalot. To achieve this you will need to have correct NAT settings in your ATA. I am not familiar with the model ATA you have, so I can't suggest any specifics. In your Voxalot account, try to avoid using the symmetric NAT option, and for outbound providers try to use the Optimize Audio Path option.
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Unread 04-04-2009, 07:16 PM   #5
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Quote:
Originally Posted by boatman View Post
Look at the codec settings in your ATA configuration. To be sure of using just one codec (for testing purposes), disable the others.
I will try...

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Originally Posted by boatman View Post
The audio packets may flow better if they are not coming through Voxalot. To achieve this you will need to have correct NAT settings in your ATA.
I am not familiar with the model ATA you have, so I can't suggest any specifics.
Well, now i'm not using the ATA as router, my network configuration is in this picture.

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In your Voxalot account, try to avoid using the symmetric NAT option, and for outbound providers try to use the Optimize Audio Path option.
I try that, but i lost the inbound audio with this configuration.

Well, today the voice is working fine (without any changes) i don't know why yesterday i have some problems.

I thinking in buy this: PRODUCTOS - TECNOERA.com
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Unread 04-05-2009, 02:25 AM   #6
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This may be the cause of the intermittently broken audio; your ICSG302 SIP unit is giving your LAN address as the address for media stream connection.
"Connection Information (c): IN IP4 192.168.1.90".

When your RTP connection address is a LAN address Voxalot may think a direct RTP connection will fail, Voxalot may maintain the RTP stream through a Voxalot server, a direct RTP connection may not happen as expected. Check that your chosen STUN server is working, or find some other way to inform the ICSG302 of your public IP address.

Last edited by boatman; 04-05-2009 at 02:28 AM.
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