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Highlighted
Beginner

Why I need Media termination point? CUCM 10.0.1

Scenario:

1-) Phone1 calls a PSTN phone number (5555555)

Phone 1 (ext1001)-----SIP---->CUCM----SIP--->CUBE--- SIP-->PSTN

2-) Call is established and phone 1 initiates a transference to another PSTN number (6666666)  putting 5555555 on hold

2-) phone 1 establishes the call with 666666 and when the transfer button is pushed:

              a) with media temination point checked in the CUCM sip trunk to CUBE the transfer works and the call between 5555555 and 6666666 is                           connected.

              b) without media termination point checked the transfer works and the call is connected but with no audio, just dead air.

 

Can somebody explain me how this comunication works? How is the call flow? Why I need MTP required checked?

 

I would aprecitte any help. Thank you in advance.

13 REPLIES 13
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Hall of Fame Master

There are several reasons MTP can get invoked as described in the SRND, etc.  Most common ones are:

DTMF mismatch

Early Offer negotiation (i.e. not supported on gen1 phones)

Supplementary features not supported 

To figure out why MTP is invoked you'd need to pull CCM logs and search for the call and reasons MTP is invoked.

 

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I'm trying to get pertinent logs with RTMT but with no luck.

would you help me in how to get these MTP logs?

thank you.

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Were you able to resolve this?   I am having the exact same problem.

 

I do not want to use "MTP Required" as it disables video.

 

thanks

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Actually I found the fix in another thread.

 

On CUCM go to System--- Service Parameters and look for "Duplex Streaming Enabled" set this to TRUE and restart the CCM service

https://supportforums.cisco.com/discussion/12262941/one-way-audio-after-call-put-hold

 

 

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Rising star

can you share the CUBE config?

also debug ccsip message for a failed call (MTP Unchecked) and a working call (MTP checked)?

//Suresh Please rate all the useful posts.
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Yes I can. They are are attached below

Called numbers are 6666666 and 7777777.

Thanks

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Hello,

 

does this issue occur when SCCP phones registered in CUCM trying to do the call transfer?

 

in the SIP profile applied to the involved SIP phones, RFC 2543 Hold is checked under 'Parameters used in Phone'? if so, can you uncheck and try once?

 

If RFC 2543 Hold is unchecked, then in the SIP profile applied to SIP Trunk to CUBE, can you check "Send send-receive SDP in mid-call INVITE" and check the behavior?

 

All these to be tried with MTP unchecked in the SIP Trunk.

//Suresh Please rate all the useful posts.
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If "Send send-receive SDP in mid-call INVITE" does not do the trick then you can set this option "Duplex Streaming Enabled" to True from the service parameter it will make CUCM to send a=sendrecv.

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didn't work. 

thanks

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Sorry for  late answer

I tried with MTP unchecked

 

RFC 2543 hold is unchecked, "send send-receive SDP in mid call INVITE" didn't work

Thanks

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Beginner

Too old Post :)
I've heard in the videos by Jeremy Cioara that if you are on call with PSTN number and if you use any of Transfer or Hold or Conference; it will initiate MTP and voice gateway router does that job.
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Hall of Fame Cisco Employee

No, that's not quite right, *ideally* you should not need any MTP to bridge the connections, and the one that handles the intelligence to allocate it, would be CUCM, not the GW. The GW can also only be used as MTP if it has been configured and registered as an MTP resource.

HTH

java

if this helps, please rate
Highlighted

Correct. So if Router HW resources are configured in CUCM at the first place under MRGL having media resources selected from router then in that case MTP comes in to the picture. Ideally CUCM resources are in first place and then router resources comes after.