Hi,
I need some explanation for Call Flow and Codecs selection (with hardware based resources).
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.