G729 codec support?

Getting started with the SIP Sorcery
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majo
Posts: 2
Joined: Mon Jul 23, 2007 6:31 pm

G729 codec support?

Post by majo » Mon Jul 23, 2007 6:35 pm

Hi MySIPSwitch Team,

Which level of G729 codec support is available. Passthrough or compelete transcoding if needed?

Regards.

majo

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Jul 24, 2007 10:10 am

Hi majo,

The sipswitch does not deal with media at all so the ability to use g729 or any other codec depends upon the destination SIP server you are sending the call to.

Hth,

Aaron

majo
Posts: 2
Joined: Mon Jul 23, 2007 6:31 pm

Post by majo » Wed Jul 25, 2007 9:10 pm

My provider support only G729 codec & my fritzbox ATA does not have this codec. So it mean I can 't use MySipSwitch.

Anyway thanks.

majo

pablo
Posts: 54
Joined: Thu Jul 12, 2007 11:01 pm

Latency due to registration in Sipswitch

Post by pablo » Thu Jul 26, 2007 6:10 pm

Aaron wrote:The sipswitch does not deal with media at all so the ability to use g729 or any other codec depends upon the destination SIP server you are sending the call to.
Aaron, does the VoIP traffic go through Sipswitch or just get redirected to the contact address?
If Sipswitch only redirects the traffic to the contact address, would it be correct to assume that there is no added latency due to Sipswitch ?
In actual fact, I have not noticed any added delay, but that is a subjective observation.

Thanks

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Re: Latency due to registration in Sipswitch

Post by Aaron » Mon Jul 30, 2007 4:19 pm

pablo wrote:
Aaron wrote:The sipswitch does not deal with media at all so the ability to use g729 or any other codec depends upon the destination SIP server you are sending the call to.
Aaron, does the VoIP traffic go through Sipswitch or just get redirected to the contact address?
If Sipswitch only redirects the traffic to the contact address, would it be correct to assume that there is no added latency due to Sipswitch ?
In actual fact, I have not noticed any added delay, but that is a subjective observation.
Hi pablo,

If you mean latency in the audio - as oppossed to post dial delay or some other signalling aspect - then the sipswitch is not involved at all. It never sees an RTP (audio) packet and would not know what to do with one if it did. The audio is always directly between your client device and the provider the sipswitch has passed the call onto. That makes it impossible for the sipswitch to have any affect on any aspect of the audio.

Hth,

Aaron

User avatar
TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Tue Nov 13, 2007 8:01 pm

majo wrote:My provider support only G729 codec & my fritzbox ATA does not have this codec. So it mean I can 't use MySipSwitch.

Anyway thanks.

majo
wrong, you can't use it with your Fritzbox ATA, at MysipSwitch, they don't have a magic wand :)
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

User avatar
TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Wed Nov 14, 2007 7:35 pm

I guess MySIPSwitch also has no influence on the address translation, or can't have any influence ?
Only the begin and end point ?
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

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