Implementazione conferenza/join softone

Hy
i am implementing a SIP softtone. I have implemented almost all the features: obviously called, waiting, blind or attendent transfer. What is missing is the join of the calls or rather I can’t understand how it works. From what I have seen, on other phones, the join takes place with an invite / reinvite that asterisk seems to translate into a left join on all conference phones. Up to now everything is ok, only that the packets arrive to me from all the conference sources (I have noticed this by sniffing the packets that pass and I have seen the different Ssrc field for each RDP packet coming from the various phones).
In this way the audio quality is very bad as I don’t correctly reconstruct the sequence of the packets I receive coming from all the phones simultaneously.
Now the question, sorry if I dwelt, this is how asterisk really manages conferences, sending all the packets indistinctly to all the phones and then it is the phones that have to merge the audio packets together (which is not implemented in java and so I’m looking on the internet) … or I’m doing something wrong in compiling the SIP request to be sent to asterisk, so that he is the one to merge the RDP packets so as to send a single RDP stream (single Ssrc) to all the phones in call?

thanks in advance Massimiliano
PS I hope I was clear

You could let Asterisk do much of that work! See the ConfBridge application:

https://wiki.asterisk.org/wiki/display/AST/ConfBridge

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