[cisco-voip] OT: Call Flow and Codecs

[Syed Khalid Ali]

Hi,
I need some explanation for Call Flow and Codecs selection.

I have the following scenario.

[Analog-Phone-1]<--telco-->[Voice-Gateway(G.711)]<--->[CUCM]<---->[IP-Phone-1(G.711)]
                               |
          |
                               |
          |-----[IP-Phone-2(G.711)]
                               |
          |-----[IP-Phone-3(G.729)]

                  [Analog-Phone-2]

Case-1: Analog Phone-1 call to IP-Phone1. Call received by VG, since both
the phone and VG are capable to handle G.711, call should be successful.Call
should also be if initiated by IP-Phone-1.

Case-2: Analog Phone-2 call to IP-Phone-3. Call received by VG, call should
be successful if a Trancoder is provided, since IP Phone-3 is using a
G.729. Call should also be successful if initiated by IP-Phone-3.

Case-3: Call initiated by IP-Phone-1 to Analog-Phone-1, IP-Phone-3 must be
included in conversation so conference is initiated from IP-Phone-1 to
include IP-Phone-2. Conference Bridge is required to successfully
include IP-Phone-2.

Case-4: Analog Phone-2 call to IP-Phone-3. Call received by VG, call should
be successful if a Trancoder is provided. At the same time, if IP-Phone-1
and 3 to be included in the conversation, a Conference Bridge is also
required. This should hold true if call was initiated by IP-Phone-3.

Case-5: Analog Phone-2 call to IP-Phone-3. Since codec are different,
 Transcoder is required to successfully complete the call. Now IP-Phone-3
put Analog-Phone-2 ON-HOLD. To provide these supplementary services a
Hardware based MTP is required to compensate the codec mis-match.

Case-6: IP-Phone-2 calls IP-Phone-3. The call is successful if Transcoder is
provide. Now IP-Phone-2 need to transfer this call to IP-Phone-1. Hardware
based MTP is required to complete the call transfer. When the transfer is
successful, Transcoding resource should be alllocated to complete the call
setup between IP-Phone-3 and IP-Phone-1.

-- 
Thanks,
Syed Khalid Ali
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