How does SIP work
Internet telephony is quickly becoming the ideal option for most firms. this is due to its increased reliability and accessibility. Two application layer protocols form the backbone for IP telephony, SIP, and RTP. The Session Initiation Protocol (SIP) exists within the Network Layer, Internet Protocol. It is responsible for the signaling in the call. Within the SIP (over UDP or TCP) the Session Description Protocol (SDP) is established. This allows the RTP (Real-Time Protocol) to be set up and helps maintain a connection for real-time content, such as voice or video. SDP is negotiated between the originating UA and the termination device (oftentimes a proxy or SBC between the actual endpoint). A session, in this case, refers to an exchange of data between two endpoints.
How does SIP function?
In terms of how it functions, the designers and developers of Session Initiation Protocol sought to come up with a call setup and signaling protocol for internet telephony. They intended for SIP to support at least as many features and call processing functions that were in the Public Switched Telephone Network (PSTN) as possible. To do that, SIP follows similar underlying concepts like its predecessor, PSTN, except it runs its calls over the internet as opposed to the traditional POTS (plain old telephone service) lines.
Using Session Initiation Protocol
To use SIP, users must first obtain a unique SIP address/account. This is provided by the chosen VoIP service provider. The user also needs SIP capable hardware or to install a SIP client on their device, also known as a ‘Softphone’. A SIP client typically contains the full-featured functionality of traditional voice lines as well as expanded features, such as Call Park.
Using the assigned SIP addresses and authentication details, in most cases, the users register with a registrar server. A user initiates the call. Then the Session Initiation Protocol sends the request to a Session Initiation Protocol server. This request includes a number or address that indicates the destination of the call. A typical call flow originating from a SIP endpoint will follow:
- First, a caller sends an INVITE request to the SIP server, which then authenticates the call and will process any call handling and call routing information.
- The SIP server will send a packet either indicating ‘Ringing’ or a failure code for varying reasons.
- The SIP server will then set up a call leg for the callee according to the routing rules for the dialed destination.
- Finally, if the destination party is available, the OL with SDP negotiated RTP establishes the call.
Does it Transport RTP?
The SIP does not itself transport RTP, but it takes care of the signaling required to establish the call. First, it helps identify user location, as demonstrated earlier. Moreover, it also helps in identifying user availability, which involves determining whether the callee is willing to engage in communication. Session Initiation Protocol also helps determine user capability. It defines which media parameters to use. It also defines what each UA can support. Most importantly, it helps set up a session by establishing session parameters at both the caller and the “callee” party. Lastly, it facilitates session management. For instance, session transfer, creating a conference, termination of sessions, and so on. It also helps in the modification of session parameters and invoking services.
SIP has had a large impact on the telephony industry, allowing for sometimes upward of 75% cost savings versus traditional premise-based POTS lines. Besides cost savings VoIP allows for increased reliability due to continuously upgraded internet infrastructure by service providers, and increased mobility of users.