Audiocodes and Brekeke

[ezzadin]

1. Brekeke Product Name and version: Brekeke SIP Server

2. Java version: 2.4.8.6/286.3

3. OS type and the version: Windows 2003 R2

4. UA (phone), gateway or other hardware/software involved: X-lite, Audiocodes

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 2 (All Phone and PBX/SIP Server are in the same LAN. You have a PSTN gateway.)

6. Your problem:

Hi All,

We currently have an Audiocodes in place to convert TDM calls to SIP. Our PBX is Dialogic based, we used a T1 cross cable to connect the Dialogic board to Audiocodes which works great. Call placed from PBX is converted to SIP and sent to our SIP provider.

Now, I want to use the same method to reach UA. So I thought of getting a SIP Registrar. I download a trial version of Brekeke SIP server, then registered the user (ext 225), I then placed a call from my PBX to user 225, he gets the call and there is audio. so the setup works ok.

I'm new to VoIP and I'm trying to figure out where the Media gets processed. I want Brekeke to act as a Registrar only and have all the Media on the Audiocodes, I think it is called Transcoding.

How can I accomplish this? Is this something on Brekeke side or Audiocodes? and what is this function called please?

Thanks

[hope]

brekekek sip server doesnot access/convert codecs.
it just relay rtp packets to avoid audio problem when caller and callee are in different network.
if caller and callee are in the same network, the rtp packets are exchanged between caller and callee without going through Brekeke server.
if caller and callee side support the same codec, the call can be established.