Site to Site CME connection G729 codec

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Sep 5th, 2008

Hi looking for some clarification, I'm aiming to create a CME to CME connection over a VPN. I presume I just point SIP dial-peers at each other but I want to use a low bandwidth codec. I guess I would need to setup codec transcoding on both CME routers? Or is there an easier way to do it?

Thanks,

Craig.

I have this problem too.
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Marwan ALshawi Fri, 09/05/2008 - 02:43

hi craig,

Transcoding is required when part of a call must use the G.711 and another part of the same call must use G.729. When you use G.729 for calls between sites, and calls forward into voice mail, these calls currently fail on the configuration, because Cisco UE voice mail supports only G.711. To fix this, configure transcoding resources on both sites to terminate G.729 calls, and transcode them locally to G.711 before they enter voice mail

SIP RFC 2833 DTMF Relay

You need SIP DTMF relay if you are using SIP trunking between sites. If you have Cisco UE integrated on your sites, as in the sample configurations built in this chapter, you must use H.323 trunking between the sites. SIP trunking is not yet supported with Cisco UE release 2.1. Note, however, that a SIP dial peer is required to route IP phone and PSTN calls to Cisco UE.

If you are using AA and voice mail solutions other than Cisco UE with Cisco CME, or a future Cisco UE software release that may support this feature, you can use SIP trunking between sites. In a SIP trunking configuration, the out-of-band DTMF relay to the SCCP IP phones must be converted to in-band RFC 2833 DTMF relay on the SIP trunk

This is done using the configuration sample bellow:

router#show running-config

dial-peer voice 2000 voip

destination-pattern 8005551212

session protocol sipv2

session target ipv4:1.1.1.2

dtmf-relay rtp-nte

good luck

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mshavrov Fri, 09/05/2008 - 06:57

You may just specify "codec g729r" on both SIP dial-peers. Your Cisco IP phones are capable to do G729 and they do not need a transcoder. However if you have any devices or services, which require G711 only (for example, CUE, B-ACD or AA TCL scripts, then you will need a transcoder at that location, and you will need strictly specify it in your dial-peers for these devices.

Good luck,

Mike

CCNP, CCVP, CCDP, CCSP, MCSE W2K, CCIE Voice (in progress)

http://www.headsetadapter.com

slongewa Fri, 09/05/2008 - 08:32

What would the transcoder configuration look like? I'm very interested in this thread, but am a bit confused by how the transcoder configuration would function.

Thanks,

Steve

Marwan ALshawi Fri, 09/05/2008 - 17:49

hi steve

transcoder in CME will be configured and used from DSPs existed in PVDMs in diffrent line cards like E1/T1

so for example u wanna configure 2 calls at time to be transcoded from diffrent codecs:

sccp local [source interface]

sccp ccm [ccm ip] identifier 1 version 4.0

sccp

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729br8

codec g729r8

maximum sessions 2

associate application SCCP

telephony-service

sdspfarm units 5

sdspfarm transcode sessions 2

with callmanager u dont need the last part which is the telephoney service instead u define the trnascoding profile in the CCM and then assign it to MRG/MRGL then to a device or device pool

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Marwan ALshawi Fri, 09/05/2008 - 17:51

so mike

based on ur post he needs this Xcoding only in case from example suer in site A want to use the pstn line in Site B right, over the WAN with g729 ?

and in case of having

CCM/CMEg711----WAN g729----rmote siteg711

in this case the phone it self can handel the transcoding without any software of hardware resources?

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