TEHO using Transcoding

Hello,

In my lab, I am seeing this:

When Site A is call Site B PSTN No, the call is routed through Site B ISDN Line (Tail End Hop Off)

The Codec used is G729r8, shown under the show voice call status and under the IP Phone Call Statistic.

Is there a way to transcode this call, so that from the IP Phone to the GW, it is using G729, but from the Gateway to the ISDN Port, it will use G711?

Reason for doing so:

In Real Life, had some issue using G729 codec with some providers ISDN Trunk, they recommend us to send G711 to them.

 

Regards,

Yen Lung 

 

Comments

  • Yen,
    This really doesn't make much sense because transcoding from g729r8 back to g711ulaw will not improve the audio fidelity at the gateway. Before the audio stream goes out onto the PRI, it will go through a DSP which will convert it to PCM anyway, no matter what codec is on the VOIP side.


    That said, if you configure a transcoder on Site B, then have your incoming dial-peer back to CUCM set to only use g711ulaw, CUCM will invoke a transcoder. At that point Site A will send g729r8 to the transcoder at Site B, then the transcoder will send g711ulaw back to itself (Site B).  This is still assuming you set your regions in CUCM so it uses g729r8 between Site A and Site B.


    Scott

    On Wed, Dec 5, 2012 at 9:04 AM, yenlung <[email protected]> wrote:

    Hello,

    In my lab, I am seeing this:

    When Site A is call Site B PSTN No, the call is routed through Site B ISDN Line (Tail End Hop Off)

    The Codec used is G729r8, shown under the show voice call status and under the IP Phone Call Statistic.

    Is there a way to transcode this call, so that from the IP Phone to the GW, it is using G729, but from the Gateway to the ISDN Port, it will use G711?

    Reason for doing so:

    In Real Life, had some issue using G729 codec with some providers ISDN Trunk, they recommend us to send G711 to them.

     

    Regards,

    Yen Lung 

     





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  • Hi Scott,

    That is what I thought too, but does not work.

    this is my partial config:

    Region of IP Phone to Region of Voice Gateway is using G729r8.

    then, on the Voice Gateway,

    I had the following Partial Config

    dial-peer voice 2000 voip
     incoming called-number .
     codec g711alaw (For E1)
     no vad
     dtmf-relay rtp-nte

    dial-peer voice 1001 pots
     destination-pattern 9.T
     port 1/0:15

    Of course, my Transcoding resource is active.

    WHen i Make a call, the Codec between the Phone and the Gateway should be G729.
    once the call hit the VOIP incoming dial-peer, it should transcode to G711. but, nothing occurs.

    WHen the Call is answer at the remote end, the call drops.

    Not sure what had happened.

  • What is your transcoder configuration? Do you have a MRGL with the included transcoder assigned to the gateway in CUCM?

    On Wed, Dec 5, 2012 at 10:06 AM, yenlung <[email protected]> wrote:

    Hi Scott,

    That is what I thought too, but does not work.

    this is my partial config:

    Region of IP Phone to Region of Voice Gateway is using G729r8.

    then, on the Voice Gateway,

    I had the following Partial Config

    dial-peer voice 2000 voip
     incoming called-number .
     codec g711alaw (For E1)
     no vad
     dtmf-relay rtp-nte

    dial-peer voice 1001 pots
     destination-pattern 9.T
     port 1/0:15

    Of course, my Transcoding resource is active.

    WHen i Make a call, the Codec between the Phone and the Gateway should be G729.
    once the call hit the VOIP incoming dial-peer, it should transcode to G711. but, nothing occurs.

    WHen the Call is answer at the remote end, the call drops.

    Not sure what had happened.




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    http://www.INE.com



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  • Both the Phone and the Gateway had a Transcoding resource allocate to its mrg/mrgl.

    the transcoding is configured as followly:

    dspfarm profile x transcode universal
     codec g711u
     codec g711a
     codec g729r8
     max session 4
     no shut

    when i do a debug ccsip event, (My gateway is configured as a SIP Gateway), i saw that the codec is negoiated is none.

    however, when I insert the voice class codec into the incoming voip dial-peer.

    the codec negotiated is g729r8.

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