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