Hi, The signalization protocol used to make, answer, or hangup calls is SIP (RFC 3261). The RTP session is used to send/receive media (audio) (RFC 3550). The SDP session is used to negotiate codec to be used during the RTP session (RFC 4566).
Hope this will be usefull. Regards, Alexandre ----- Original Message ----- From: [email protected] To: [email protected] Sent: Friday, October 29, 2010 11:45:34 AM GMT -05:00 US/Canada Eastern Subject: [SFLphone] [Technical] PHP Sipphone james sent a message using the contact form at http://www.sflphone.org/contact. Hi, Hope you can help me out. Am trying to develop a web-based sip phone based on PHP and Opensips 1.6.5. Am abit stuck in the implementation and I was hoping if you could help me with some guidance. Basically am able to do a 3 way handshake and connect 2 parties using a proxy. My challenge is then initiating a call and getting audio from one user to another. From what I've read i need to use SDP and RTP. Please help me and give me more guidance on what I need to do need. Thank you. regards, James _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
