Basic VoIP concept: Overview of the SIP protocol

SIP is an application control signaling protocol proposed by IETF. As the name implies, it is used to initiate a session. It can be used to create, modify, and end multimedia session processes attended by multiple participants. Participants can communicate with each other through multicast, unicast, or network connection.
There are clients and servers in the SIP. A client is an application that establishes a connection with the server to send requests to the server. The User Agent (User Agent) and Proxy) contain clients. A server is an application used to provide services and send responses to requests sent from a client. There are four types of basic servers:
· User proxy server: When a SIP request is received, it contacts the user and returns a response on behalf of the user.
· Proxy server: initiates requests on behalf of other clients, acting as both a server and a media program of the client. Before forwarding a request, it can rewrite the content in the original request message.
· Redirect server: it receives the SIP request and maps the original address in the request to zero or multiple new addresses and returns them to the client.
· Registration server: it receives client registration requests to complete user address registration. User Terminal programs usually need to include user proxy clients and user proxy servers. Proxy Server, redirection server, and registration server are public network servers. The concept of locating a server is also often mentioned in SIP, but locating a server does not belong to the SIP service.
SIP fully considers the scalability of other protocols. It supports many address descriptions and addressing methods, including username @ host address: callednumber @ PSTN gateway address: Tel: 010-62281234 common phone descriptions. In this way, the SIP caller can identify the location of the called phone on the traditional telephone network according to the called address, and then initiate and establish a call through a gateway connected to the traditional telephone network. The most powerful feature of SIP is the user positioning function. SIP itself contains the registration function to the registration server. It can also enhance its positioning function by using the positioning services provided by other locating server DNS and LDAP.
SIP provides six types of signaling: INVITE, ACK, CANCEL, OPTIONS, BYE, and REGISTER. INVITE and ACK are used to establish a call, complete three-way handshake, or change session properties after the call is established. BYE is used to end the session. OPTIONS is used to query server capabilities; CANCEL is used to CANCEL a request that has been sent but has not ended. REGISTER is used to send a message such as a user location registered with the registration server.


Figure 6-8 example of creating a call using a SIP proxy
The SIP protocol supports three call methods: the user agent service machine UAC directly calls the user agent server UAS, UAC redirects the call with the assistance of the redirection server and the proxy server initiates the call to the called call on behalf of UAC. Example 6-8 shows how to create a call using a SIP proxy.
Comparison of H.323 and SIP protocols
H.323 and SIP are suggestions for the communication and Internet camps. H.323 attempts to regard IP phones as well-known traditional phones, but the transmission mode has changed from circuit switching to group switching. The SIP Protocol focuses on using IP phones as an application on the Internet, which has higher requirements for signaling and QoS than FTP and E-mail, they support basically the same business, and also use RTP as the media transmission protocol. But H.323 is a relatively complex protocol.
H.323 uses a binary method based on ASN.1 and compression encoding rules to represent its messages. ASN.1 special code generators are usually required for lexical and syntax analysis. The text-based protocol of SIP is similar to HTTP. Text-based encoding means that the meaning of the header domain is clear at a glance, such as From, To, Subject and other domain names. This distributed, almost no complex document describes the standard and standard f style, its superiority has been proven in the past. SMTP is a popular mail protocol ). The message body of the SIP is described using SDP. The format of each item in SDP is =, which is also relatively simple.
In terms of support for conference calls, as H.323 is centrally implemented by the multi-point control unit MCU), all participating Conference terminals send control messages to the MCU, And the MCU may become the neck, especially for large-scale conferences with additional features, and H.323 does not support the multicast function of signaling, its single function limits scalability and reduces reliability. The Design of SIP is a distributed call model with the distributed multicast function. The multicast function not only facilitates conference control, but also simplifies user positioning and group invitation, it also saves bandwidth. However, the H.323 set is in medium control to facilitate billing, and bandwidth management is relatively simple and effective.
Special protocols are defined in H.264 to supplement services, such as h.2.1, h.2.2, and H.264. SIP does not specifically define a protocol for this purpose, but it easily supports supplementary or intelligent services. You only need to make full use of the header domain defined by the SIP, such as the Contact Header domain, and make simple extensions to the SIP, such as adding several domains) to implement these services. For example, for call transfer, you only need to add the Contact Header domain to the BYE request message and add the third-party address to be transferred. For some intelligent services that are difficult to implement by extending the header domain, you can add business agents in the architecture to provide some supplementary services or interfaces with intelligent network devices.
In H.323, the call establishment process involves the third signaling to: RAS signaling channel, call signaling to and H.245 control channel. Through the coordination of these three channels, the H.323 call can be carried out, and the call establishment takes a long time. In SIP, Session Request and media negotiation are performed together. Although H.323v2 has improved the call establishment process, it still cannot be compared to that of SIP, which only requires 1.5 loop latency to establish a call. H.323 call signaling channels and H.245 control channels require reliable transmission protocols. The SIP protocol is independent of the Low-layer protocol. Generally, the protocol such as UDP cannot be connected. The reliability mechanism of the credit layer is used to ensure reliable message transmission.
In short, H.323 follows the traditional telephone signaling mode, which is relatively mature and has already seen many H.323 products. In line with the traditional design concept in the communication field, H.323 implements centralized and hierarchical control and uses the H.323 Protocol to facilitate connection with traditional telephone networks. The SIP protocol draws on the design ideas of other Internet standards and protocols and follows the principles of simplicity, openness, compatibility, and scalability that the Internet has always adhered, however, the launch time is not long and the Protocol is not very mature.