SIP Router
Yate can be used as a SIP router with his features like registering, proxy and redirect just to mention a few. The implementation of SIP it is done using a VoIP SIP driver based on YASS library.
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What's SIP?
SIP comes from Session Initiation Protocol.
SIP is used to negotiate a call between 2 Endpoints. SIP negotiate the:
- media - text, voice or other type of media
- transport - UDP, TCP, RTP
- encoding
Once SIP call is established, the communication take place over the specified transport mechanism independent of SIP.
What does this definition mean in our case? This means SIP it is used to initiate and maintain sessions in a network to provide VoIP communication between two or more points. Such points could be one phone and one PC for example or more computers located in various places in case of a multi-conference, phisycally connected in such a way that VoIP communication to be possible.
Yate as a SIP Router
There are various protocols used to carry data over a real time multimedia session in numerous formats like voice, video or text messages.
SIP works along with them, helping the endpoints in a communication to agree about the session characteristics.
Other protocols used in session communication
SIP doesn't offer any other services besides the session configuration related ones, that is why is being used along with protocols specialized on other tasks like the:
- Real-time Transport Protocol (RTP) for data transporting and Quality of Service;
- Session Description Protocol (SDP) for describing multimedia sessions and others.
This way SIP can provide complete functionality to the users but despite all this collaboration with other protocols it does not depend on them.
As a SIP router Yate supports this protocol and its features like registering, proxy and redirect, through its own library that implements them, this being called YASS – Yet Another SIP Stack.
Features that an implementation of SIP permits
Besides establishing, modifying and finalizing a session, SIP can also be used to invite participants to an already existing one.
Also a type of media can be added or removed from a session. Like stated in the RFC document 3261 which represents the last published standard for SIP, the protocol supports name mapping and redirection services and users can maintain a single externally visible identifier regardless of their network location.
The call forwarding SIP provides to the implementing servers is accompanied by the possibility of negociating the terminal type and capabilities and selecting it. This way a caller is given a choice about how to reach the party, via Internet telephony, an answering service, etc.
The security of communication services always was considered important so SIP provides a suite of security services like user authentication, denial-of-service prevention, integrity protection, and others.
See also
