I am looking for a mechanism for audio streaming in my network. I will be using server for transmitting audio data in RTP format and standard players such as VLC for receving and playing data. As my data format is not a static RTP payload type, I am planning to use a SIP server for transmiting SDP information(for sample rate, no of channels, sample width etc). Can I use 'opensips' stack for this purpose? please guide me.
OpenSIPS is not a SIP stack, but a fully sip Server - and you cannot extract only the stack.
But you can use opensips to initiate calls (with your SDP) via the management interface (using t_uac_dlg - see http://www.opensips.org/html/docs/modules/1.7.x/tm.html#id294568)
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