The Session Initiation Protocol (SIP) is a protocol for controlling a communication session between two or more participants. It is specified in RFC 3261. The protocol is modeled on the Hypertext Transfer Protocol and is an open, well-extensible standard that is commonly used for example in the IP telephony.
SIP is used to negotiate the communication itself. The actual data exchange is realized via separate protocols. Often this is the Session Description Protocol (SDP, RFC 4566) embedded in SIP, to negotiate details on the video and/or audio transmission. Here, the possibilities of the communication partners are taken into account, i.e. which codecs (transfer methods) they can support and with what protocol they want to transmit the data.
SIP provides flexibility.
SIP controls the communication controller. A separate protocol, such as SDP, agrees the details of data exchange. The actual data is then transmitted with a transmission protocol. Since the individual components has been working together, but are only loosely connected, the components are interchangeable. A manufacturer could, for example, create a custom protocol to clarify the data exchange and/or use its own data transfer protocol. Still, he could draw on its SIP server infrastructure.
Another advantage is, that an existing connection can be changed. Communication channels and subscribers can be added or removed.
Internet Telephony
SIP is often used for Internet telephony. The embedded SDP in SIP negotiates the transfer details. The actual transmission realizes in this case the Realtime Transport Protocol (RTP, RFC 3550, RFC 3551) or Secure Realtime Transport Protocol (SRTP, RFC 3711).
subscriber addresses will be written in the URI format:
- Unencrypted connection SIP: sip: account @ domain.
- Encrypted SIP connection: sips: account @ domain.
- Telephone: tel: number, for example, tel: +49-30-1234567.
Since the RTP uses dynamically allocated UDP port, problems may occur in conjunction with the firewalls that has to open the required ports as required. Also it can lead to problems in the Address translation. Supporting tools and protocols help to avoid this, so that SIP-based IP telephony also works with routers, NAT and firewalls.
