I am thinking of making my own VOIP client in Rust. Therefore, I had a look upon SIP protocol:
The question is though how I can handle upon a sucessfull login and call acceptance the RTP protocol. The question is on a typical VOIP client is a single socket used for both SIP and RTP or a single socket is used instead?
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In a typical VOIP client, a separate connection is used for the RTP protocol. The SIP protocol is used for signaling, which involves establishing, modifying, and terminating sessions that involve video, voice, messaging, and other communications services. The RTP protocol is used for the actual transmission of the audio or video data.
Because the SIP and RTP protocols serve different purposes, they are typically handled on separate connections in a VOIP client. The SIP protocol is used to establish and manage the session, and the RTP protocol is used to transmit the audio or video data.
To implement a VOIP client in Rust, you will need to establish two separate connections: one for the SIP protocol and one for the RTP protocol. You can use the
sip,sdp, andrtpcrates to handle the SIP and RTP protocols in your Rust client.I hope this helps! Let me know if you have any other questions.
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