Hi All! I'm trying to build a proof-of-concept mobile app that will involve VOIP functionality. VOIP is not core functionality to my idea, but it is an important feature. I was hoping y'all could test my assumptions and point me in the right direction. :-)
Background: Of course, I have no VOIP experience, and limited coding skills but I've got a strong IT architecture background. I believe in building cross-platform as much as possible. I believe strongly that how you start is how you finish, so rather than just bash code I'm starting with locking down my n-tier architecture. The idea is to write a decent spec, and find some decent freelance coders to help me implement my POC (I would be paying them out of my pocket). Since I've got no cash to speak of (I'll be funding this proof of concept out of my normal income) to keep capital requirements low I'll use AWS where-ever possible. Use-case: The idea is that my users will want to communicate with other, either one-on-one or many-to-many. There would be a "lobby" of other users filtered by criteria set by the user. The users could form a group and start a conversation or join a conference in progress. There would be a buddy list too. Current thoughts: When in "lobby" mode, the app would be communicating with a lobby-server which would act as a directory of some kind. My understanding is that Linphone requires a SIP server to perform multi-party conferencing. The preference would be to have each client establish a stream up to a server and for the return stream to contain a muxxing of all other participants streams. This way the bandwidth requirement for the user's device would be more or less the same for one participant or multiple participants. When a conference is initiated, the "lobby" server should establish a new conference on the conferencing server- it would then send the details back to the user's client and then it would establish a connection to the SIP server and join the conference. Questions: * Is Linlibphone the right library for the voice-transport part of this job? Is there something more suitable? * What open-source SIP servers can perform the muxing & conferencing hosting? * Can the VOIP stream be tunnelled over SSL? * What about functions like Moderator (kicking people out of conference, etc) - would this be performed on the SIP server? Is there an appropriate library I should look at? * What about noise suppression and stuff like that? * What do I need to consider that I haven't? Thanks for your time - I really appreciate any thoughts or wisdom you could offer :) -James
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