Solved: VOIP between Cisco Voice Gateway

lumguanyuan · ‎03-24-2009

3 X Cisco 2821 configure voice gateway for 3 different sites. Each voice gateway have a voip dial peer to another sites and vice verse.

Let say all 3 sites 3rd party PBX and support E1 truk to Cisco voice gateway. Only use Cisco voice gateway route call to different site.

How can we do the transcoding, MTP and conference call based on this setup? Or this kind of setup only use G711codec?

Please help.

orochi_yagami · ‎03-24-2009

Hi Lum,

What Paolo means is that you can configure the codec preference,then apply it under dial-peer.

View solution in original post

gopinath.j · ‎03-24-2009

Hi,

Voicegateway does not have the inteligence to do conference.Can be achieved only with Callmanager functionality.

You can use any supported codec on the dialpeer.

View solution in original post

Nicholas Matthews · ‎03-24-2009

A voice gateway in the traditional sense will only route calls. Every call will require DSP resources, because DSPs are needed to translate non-RTP voice on analog circuit to packetized RTP voice. Some codecs require larger amounts of DSPs, because for a single DSP you may be able to place 16 G.711 calls but only 8 G.729 calls. In this function, all calls are independent of one another, and there is no controlling manager.

For things like MTP, conferencing, transfers, and transcoding, it requires for multiple streams to interact. For interaction, it requires a CME or CUCM/CCM.

Once you have a CUCM/CME, you can have something where there are 8 gateways, but only 5 of them have the DSPs for conferencing and 2 do transcoding, etc.

hth,

nick

View solution in original post

paolo bevilacqua · ‎03-24-2009

You can use any codec you want, just configure under DP. Recommend you use G.711 for best quality and ease of confiuration.

For conferencing, recommend you configure one or more of the routers as CME, then configure conferencing as explained in the system administrator guide. By calling the specified number from any site, you would be conferenced in.

lumguanyuan · ‎03-24-2009

Let say the router just a voice gateway and no CME. Can conference be achive?

What is DP you mean for?

orochi_yagami · ‎03-24-2009

Hi Lum,

What Paolo means is that you can configure the codec preference,then apply it under dial-peer.

lumguanyuan · ‎03-24-2009

Orochi Yagami,

I need DSP if I want to use G729 right?

How about conferencing?

gopinath.j · ‎03-24-2009

Hi,

Voicegateway does not have the inteligence to do conference.Can be achieved only with Callmanager functionality.

You can use any supported codec on the dialpeer.

lumguanyuan · ‎03-24-2009

Hi gopinath.j,

can I say it supported for call route only based on my scenario.

No other function like conference and MTP.

Nicholas Matthews · ‎03-24-2009

A voice gateway in the traditional sense will only route calls. Every call will require DSP resources, because DSPs are needed to translate non-RTP voice on analog circuit to packetized RTP voice. Some codecs require larger amounts of DSPs, because for a single DSP you may be able to place 16 G.711 calls but only 8 G.729 calls. In this function, all calls are independent of one another, and there is no controlling manager.

For things like MTP, conferencing, transfers, and transcoding, it requires for multiple streams to interact. For interaction, it requires a CME or CUCM/CCM.

Once you have a CUCM/CME, you can have something where there are 8 gateways, but only 5 of them have the DSPs for conferencing and 2 do transcoding, etc.

hth,

nick