Session Initiation Protocol (SIP) is a text-based protocol, for negotiation of connections based on the Internet Protocol (IP). SIP is used merely to handle the signaling between individual negotiation parties. The transport of media data runs – similarly like by ISDN – separately from the negotiation. The media data are often sent over a different route, conveyed by some other Transport Protocol. So if there are TCP or UDP available for the negotiation, RTP is mostly used for the media data transport.
In the simplest case (Peer-to-Peer) individual components are connected directly to each other with no any central unit. Within one session the entities of one Peer are called User-Agents (UAs).
To start such negotiation there is an INVITE-message sent from the calling party to the SIP-address of the receiver. Should the receiving party accept the call, it can answer directly with an OK-message, which is then acknowledged by the calling party with the ACK-message. Each of these messages most often carries additional specific information in its so called Body where it is specified which audio and video codecs must be used, which parameters they must use and which IP-addresses and Ports have the user terminals. To describe the media data connection there is a Session Description Protocol (SDP) used.
The true mobility and ease of use for the IP-based transmissions can be reached by establishment of connection using an SIP-Server. Even if the IP-addresses of the remote units are unknown to each of the parties they can log on at some SIP-Server with a previously assigned SIP-address in form of “user@service-provider” and usually also with some password. The SIP-Server then stores the information about the current IP-addresses of the units and is able to transparently forward the same SIP-messages from one device to the other. And it doesn’t have to be one and the same Server. There can be even a chain of such SIP-proxies used.
Messages “100/Trying” shown in the Figures are optional and serve for the acknowledgment of the request, as well as “180/Ringing” messages are optional too and they solely signaling to the calling party about the ring tone at the receiver side.
More information on SIP usage for professional audio transmissions you can find in our upcoming publications in some professional broadcasting technology magazines, as well as at our Audio-via-IP Seminars to be announced in the near future at this web-site.
see also Application Note "Using MAYAH SIP Server" (PDF) » |
