DSP questions

[mike85719]

1. How are calls transcoded from G711 to G729 in a CCM environment (with remote sites separated by regions) when I do not have any transcoders configured in CallManager? CCM must be transcoding behind the scenes somewhere. Is this a non-configurable CCM software resource... or are my PVDM's on the gateways somehow being tapped even though I don't have DSP farms setup...or are the phones doing the transcoding?

2. I have a 2801 gateway with an ISDN PRI that is controlled by CCM. The calls obviously need to be converted from TDM to VoIP. Is the process of converting from TDM to VoIP configurable? I do not have a DSP farm on this gateway but it does have a PVDM installed. So I guess my question is: do all I need to do is physically install a PVDM and that?s it?

3. Can a 2801 router provide transcoding services when neither CME or CCM are involved?

For example this is the design: non-Cisco PBX -> G711 SIP trunk -> Cisco 2801 -> G729 SIP (over a WAN) -> Cisco 2801 -> G711 SIP Trunk -> non-Cisco PBX

Both routers have PVDM's but the question is do I need a "DSP Farm", and if I do, can the DSP farm be setup without CME or CCM controlling it? When you setup a DSP farm you usally need to specify the "applacation" which is usally sccp which would indicate the CCM or CME needs to be involved.

[mchandak]

With TDM to VoIP Calls, the converstion happens on the PRI interface. When a PRI is brought up by the Gateway it assigned PVDM to these channels and therefore they can handle the conversation from TDM to VoIP and visa-versa. Only option you can configure here is which codec do you want the call to use under the voip dial-peers.

With the senerio you have mentioned above, you will need to configure Transcoder as the we are not doing a TDM to VoIP, we are doing VoiP to VoIP which can only be done forcing a transcoder as by default there is no auto mechanism to do this on Cisco Routers as VoIP does not need DSP resources by default.

A Transcoder can only be configured if u have a CCM or CCME as a transcoder works more like a Skinny device which can only be controlled by a CCM or CCME

[paolo bevilacqua]

Mike,

in point 3, what are you wanting to do, requires an ip-to-ip gateway image, and yes, you should be able to have transcoding this way, without CCM or CCME.

See: http://cisco.com/en/US/products/sw/voicesw/ps5640/products_data_sheet09186a00801da698.html

Transcoding G711 a-law and mu-law to G.729, G.729a, G.729b, and G.729ab**

Hope this helps, if so please rate post using the scrollbox below!

[mike85719]

Thanks for the replies.

I found another link that says that "Transcoding is not supported" on SIP to SIP connections:

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_guide_chapter09186a00806b0a14.html

However it is supported on H323 to SIP connections. So I should be able to change my design from this: non-Cisco PBX -> G711 SIP trunk -> Cisco 2801 -> G729 SIP (over a WAN) -> Cisco 2801 -> G711 SIP Trunk -> non-Cisco PBX

To this: non-Cisco PBX -> G711 SIP trunk -> Cisco 2801 -> G729 H323 (over a WAN) -> Cisco 2801 -> G711 SIP Trunk -> non-Cisco PBX

Any reason why this wouldn't work? I haven't been able to find any documentation on how to setup the transcoding resources on the IPIPGW. Is this something that just works if the transcoding resources are physically installed on the router or do I need to configure a DSP farm?

[paolo bevilacqua]

Hi Mike,

I haven't found the reference to "Transcoding is not supported" on SIP to SIP connections in the reference document document. To the contrary, it says it should be supported:

SIP-to-SIP Basic Functionality for Session Border Controller (SBC)

...

* Codec transcoding

In any case, it is my understanding that you only need sufficient codec resources (PVDMs) in the router, and no dspfarm configuration is necessary.

I will give this a try on my router as time permits.

Thank you for the nice rating and good luck!

[mike85719]

Thanks for clarifying the PVDM/DSP Farm issue.

In the document, I was referring to: "Restrictions for Configuring SIP-to-SIP Connections on an IP-to-IP Gateway" -> "Transcoding is not supported".

I now see where it says transcoding is supported under "SIP-to-SIP Basic Functionality for Session Border Controller (SBC)". I guess it's a little misleading.

[mmorris11]

Mike,

The feedback on this thread is solid. However, I think you might be in need of clarification of something very fundamental in Cisco IPT. It could be that since you may have region A set up as a g711 and region B set up as g729a that calls between those sites appear to be "magically" transcoded without the necessary hardware. This is not the case. What the region does is select a default codec for a device to use based on whether the call destination is inter-region or intra-region. Thus when a phone in region A calls a phone in region B, the codec used for that call will be g711 end to end. The region B phone will accept the call and use g711 simply because the code running on the phone supports it. Reverse the scenario and g729 would be used for the same reason. Now if while the former call were in progress and a third party from region B were to call the region A phone we might have a problem for a conference. The outgoing call would be g711 but the new incoming call would be g729 (the default codec for region B) and the conference would fail, without transcoding resources. Callmanager can provide software transcoding via a meet-me conference, or you could implement hardware dsp resources for the ad hoc conference.

HTH

-mike

[mike85719]

Thanks for clarifying my region question. I think the key phrase is "end-to-end". It makes sense now.