Doubt about transcoding

[wilsonsant]

Hi,

We are implementing the IPCCX version 8.0.2 and I am the following doubt: For all inbound call that are forward to CTI Route Point (IPCCX) is open the transcoding session. I would like know it is possible not is use the transcoding. If there this possibility, I need that DTMF could be forward the correctly to IPCCX. Is there the possibility?

Thanks,

Wilson 

[Abu Hadee]

Hi Wilson,

Your question is not clear to me. UCCX doesn't invoke transcoding. It's CUCM who invokes transcoding depeding on the codec on two leg. So, if you have G711 all the way, then you do not need any transcoding.

Now, depending on DTMF Relay, CUCM may invoke MTP if it sees two different DTMF on two leg of the call.

Please clarify a bit more what you need, what is the call flow etc.

Thank you

- abu

[wilsonsant]

Hi Abu,

Below is my scenario:

IPCCX Environment:

·         The calls to arrive in the ITSP  by CUBE that forward the calls to CUCM:

ITSP ß SIPà CUBE ßSIPàCUCM

                - Configuration Dial-peer in the CUBE:   

                                                                       dial-peer voice 100 voip

translation-profile incoming coloca-prefixo-SPO

destination-pattern  xxxxxx....

session protocol sipv2

session target ipv4:x.x.x.x

incoming called-number xxxxxx....

voice-class codec 1

voice-class sip early-offer forced

dtmf-relay rtp-nte

no vad

·         The Calls enter in the CUCM and is associate with CTI Route Point to IPCCX;

·         The IPCCX is configured to work with codec g729;

·         The Region of  IPCCX is configured to work in codec  g729 with another Region configured in the CUBE;

·         O SIP TRUNK is configured to not work MTP;

·         O SIP TRUNK is configured to work with DTMF RFC2833;

Unity Environment:

·         The Calls arrive of the ITSP by CUBE that forward the calls to CUCM:

  ITSP ß SIPà CUBE ßSIPàCUCM

                - Configuration Dial-peer in the CUBE:

                                                                       dial-peer voice 100 voip

translation-profile incoming coloca-prefixo-SPO

destination-pattern xxxxxx....

session protocol sipv2

session target ipv4:x.x.x.x

incoming called-number xxxxxx....

voice-class codec 1

voice-class sip early-offer forced

dtmf-relay rtp-nte

no vad

·         The Calls to enter in the CUCM and is forward to UNITY by VOICE-Mail Ports;

·         Unity can work codec g729 ou g711;

·         Region of  Unity with Region do SIP TRUNK is configured to work  with g729.

·         O SIP TRUNK is configured to not work MTP;

·         O SIP TRUNK is configured to work with DTMF RFC2833;

The transcode is using in this solution because is implemented G711 to call internal and G229 to external calls.

Thanks,

Wilson