What factors determine synchronous signal transmission ability of a channel? So I am Designing a system which would be able to send and receive stereo voice signals.( like a telephone but with full-band stereo voice signals)

here is an example to make my question clearer. 
In a music recording studio two microphones are foreseen, one for singer's voice and the other for the guitar's sound. The combination of these two voice signals will result into a Stereo signal.At the same time there is a live conversation with another end which is supposed to receive music studio's signals as stereo and be also able to send back stereo signals.
If Bluetooth were to be used to transmit voice signals to CE Device( smartphone) and VoIP to transmit between two sides of conversation . I have done some studies but apparently Bluetooth has some limitations in case of synchronous transmissions.
what factors should be taken into account to choose a transmission method?
Not sure what exactly you mean by "synchronous". Most digital transport mechanism (such as Ethernet or 802.11) are "asynchronous", i.e. a continuous data streams  are chopped up in packets, the packets are transmitted whenever the connection allows it, the packets are assembled in the correct order at the receiving end and then turned into a continuous data stream again. Since it's not known when exactly the packet can be transmitted, there has to some buffering at the receiving end so there is some data available even if the next packet comes in late. This results in a certain amount of "latency". In the case of Bluetooth + VoIP you are probably looking at several 100 milliseconds to a few seconds of overall latency.

You may also be concerned about inter-channel sync, i.e. do both channels of the stereo stream stay time aligned. This can easily be achieved by interlacing the channels to create a single stream.

This all depends heavily on the specific implementation: Bluetooth supports many profiles. For your application you probably want the A2DP profile. This uses the SBC codec and there are dozens of ways to configure the codec ranging anywhere from high quality to totally unusable. Both parts of the interface need to agree on the codec implementation.

There are also many different VoiP protocols and codecs out there. Most operate in the 10kb/s-40kb/s range which seems way too low for you application, so you may need to create a custom protocol.