Happy to announce that our first participation to Google Summer of Code with Kamailio Project shows the results.
Marius-Ovidiu Bucur, our selected student, succeeded to blend properly his resources to take care of University exams, job and GSoC project.
He has just pushed the code for SIMPLE presence server for conference calls (RFC 4353 and RFC 4575) into public GIT repository, for now it resides in branch: mariusbucur/conference
It comes along with a testing suite, being the first implementation in an open source SIP server I am aware of, there are not many options of clients and media servers that can be used. However, this is how SIP Express Router (SER) and Kamailio (OpenSER) pushed forward many times SIP and VoIP market over the years -- bringing innovation, offering to others necessary framework to start implement by themselves new features or new services.
SIP is a client-server communication protocol, facing many times the chicken-egg problem (i.e., there is no client implementation supporting extension X because there is no server implementing X, which is not there because is no client...). We take pride of unlocking many such situations in the past (e.g., IPv6, TLS, SCTP).
In this particular case, the project started upon suggestion of our colleagues from SIP Communicator project, which has already implemented these specs from SIP client point of view. What is still missing is a media server (audio conference mixer) offering these features. Looking forward to Asterisk, FreeSwitch or SEMS.
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