I m a beginner in voice and I have multiple questions in my mind with regards to codec...
1) I wish to know whether a CODEC(g.729 or g.711 etc) is required while making a call from one IP Phone to another IP Phone(calls within one premise or remote)?
2) I have learnt that the transcoding is done in the call manager. Correct me if I m wrong.
3) I have also learnt that once the call is established the call manager is out of picture as the communication between two IP Phones is point to point and the voice traffic doesnt traverse through the call manager after the call is establshed. So for example if I establish a call from one IP Phone to another IP Phone within one premise and unplug the call manager from the network(Publisher and subscriber both) will the call get disconnected or will it remain established?
4) Some of my friends who have worked in voice say that the established call will not be dropped if the call manager is disconnected from the network however if I disconnect the call I wont be able to make a new call unless the IP Phone locates the Call manager.
5) If transcoding is done in call manager, and if suppose a call manager is disconnected in between an established call, and if the call doesn't drop then how does an IP Phone know how to transcode in absense of call manager?
In each call you have to use CODEC regardless if it is from IP phone to IP phone or IP Phone to GW ...
G.711 gives you better sound quality but more bandwidth not like G.729 so that we are using G.711 in the same region and G.729 interregions.
Transcoder is hardware and callmanger con not give you transcoding functionality you have to configure DSPs on the router to work as transcoder
Regarding the calls between the IP phones, the RTP channel is point to point and it is from IP phone to IP phone directly so that if the call manger goes down during the call, the call will not be effected.
Think of a codec as the VoIP equivalent to MP3 vs. AAC. G.711, G.729, G.722, etc are all ways of conveying real-time voice information just as MP3 is a means of conveying music. G.711 and G.729 provide roughly equivalent performance to the PSTN. G.729 uses compression to reduce the bandwidth required but comes at a cost: it cannot deal with music and some non-Latin languages because the vocal patterns do not work with the predefined word dictionary.
UCM is never out of the call entirely. Once the call completes it is true that the audio traverses directly between two phones (assuming a media resource such as a transcoder or MTP is not involved); however, UCM still must provide signaling for the call. If UCM were to become unavailable all call signaling actions (e.g. hold, transfer, etc) will stop working. You can talk and hang up, that's it.
Media resources are allocated at the begining of a call. Since you can't perform actions that would initiate a new call leg in a signaling failure scenario, you can't invoke new media resources either. The real-life experience is typically to fail over to SRST after that call ends. Transcoding is rarely required in a fail over scenario; however, it and other media resource such as conferencing can be provided by CME in SRST mode. SRST is a temporary stand-in for the phones to work with until UCM returns to service.
Cisco IP Phones by default supports G.711/G.729 codecs. Some other IP phones do support other codecs such as Cisco wideband codec/G.722 in addition to these.
The common codec is negotiated before IP phones set up the call which also depends upon CUCM Region configuration.If there are no common codecs, then the transcoding resource is invoked to translate from one to other in both directions.
For eg, let's assume that IP Phone A in HQ and IP Phone B in Branch and region configuration states to use only G.729 between HQ and branch. Then, both IP Phones communicates through G.729 though they support G.711 as per the region configurations. Transcoding comes into picture when one of the party does not support or not able to communicate with common codec due to region configuration.
Considering same example as above, if IP Phone B is using software conferencing resources of Cisco Callmanager servers in HQ and performing conferencing with others, then the call will break for IP Phone B as soon as the reachability to HQ is lost as every conferencing party was connected to software conferencing resource located in HQ for mixing and processing audio channels.
Hope it helps...
With best regards...
Pls kindly rate if helpful or answered your question.
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