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Hardware resources for transcoding, conferencing and MTP

eric.yang
Level 1
Level 1

Hi all,

 

We are using a centralized SIP trunk for PSTN access and 64kbps (G.722, G.711) for intra-region / inter-region in CUCM. The SIP provider only support g.711, do i still need hardware transcoder and hardware conferencing resources? As i understand, all internal call will be using g.722 to have better voice quality, external call will be using g.711. If there is an ad-hoc conference with a PSTN user, cucm will re-negotiate all 3 parties to use g.711. Correct?

 

MTP is not required for the SIP trunk on this case.

 

Eric

2 Accepted Solutions

Accepted Solutions

If the Region of the SIP Trunk is set to 64K with the Region of the internal phones, then CUCM would attempt to negotiate G722 first for a PSTN call and then G711 because (as you noted) the Audio Codec Preference List lists G722 'higher' making it a more desirable codec. If the gateway has no preference configured, it will negotiate G722 with CUCM and would need a transcoder resource.

To force calls from CUCM going to the PSTN to use G711 so that no transcoder is needed, you would need to set the dial-peer on the router to use G711 either by declaring that codec on the dial-peer or by configuring a voice class codec list.

And, yes, CUCM will set up a conference call hosted on a CUCM-based software conference bridge as a G711 call.

Maren

View solution in original post

Which system (CUBE or CUCM) would invoke a transcode resource would depend on the directionality of the call (inbound or outbound) and whether early offer or delayed offer is used. Set up a call and see.

However, I strongly recommend that you avoid the transcoder altogether by implementing a codec preference on the CUBE or by creating an audio codec preference list special for the calls going to/from the CUBE that sets G711 as preferred above G722.

Maren

View solution in original post

7 Replies 7

eric.yang
Level 1
Level 1

Any comments? Thanks

If the Region of the SIP Trunk is set to 64K with the Region of the internal phones, then CUCM would attempt to negotiate G722 first for a PSTN call and then G711 because (as you noted) the Audio Codec Preference List lists G722 'higher' making it a more desirable codec. If the gateway has no preference configured, it will negotiate G722 with CUCM and would need a transcoder resource.

To force calls from CUCM going to the PSTN to use G711 so that no transcoder is needed, you would need to set the dial-peer on the router to use G711 either by declaring that codec on the dial-peer or by configuring a voice class codec list.

And, yes, CUCM will set up a conference call hosted on a CUCM-based software conference bridge as a G711 call.

Maren

Thanks Maren. I have two CUBEs with HA for PSTN access via a centralized SIP trunk. We have universal transcode, conference and MTP configured on the CUBEs. If the CUBEs has no preference codec configured and g.722 is preferred codec for internal phones. As you noted, it will negotiate G722 with CUCM and would need a transcoder resource. In this case, the CUBE transcode resource will be invoked if there is no hardward transcode in the MRGL under the device pool of the internal phones? If there is CUCM registered hardware transcode in CUCM, which transcode resource will be used? 

Which system (CUBE or CUCM) would invoke a transcode resource would depend on the directionality of the call (inbound or outbound) and whether early offer or delayed offer is used. Set up a call and see.

However, I strongly recommend that you avoid the transcoder altogether by implementing a codec preference on the CUBE or by creating an audio codec preference list special for the calls going to/from the CUBE that sets G711 as preferred above G722.

Maren

eric.yang
Level 1
Level 1

One question here, if g.722 is higher than g.711 in the default Audio Codec Preference List, a desk phone only supports g.711 and another one support g.722 and g.711, g.711 will be used for the call between them or g.722 will be used with transcoder?

The audio codec preference list is just that, a list. For SIP calls, the SDP will list all of the supported codecs (in order of preference) of the offering device. The responder to the SDP will select a codec from the list that is compatible with their capabilities, based on their order of preference. A transcoder is invoked only if no common codec exists for the two devices.

Maren

eric.yang
Level 1
Level 1

Thanks Maren. Your answer has given me good insights.

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