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With the popularization of IP telephony applications, establishing a scalable network of terminal devices and gateways has become a major technical challenge for the industry. Currently, IP telephony technology has perfectly achieved the encoding and transmission of voice and fax information, but still needs to further develop call control and address management technologies for large companies and service provision platforms. Therefore, multiple protocols that support the interworking of IP telephony systems have been introduced. Commonly used IP telephony protocols such as H.323, MGCP, and SIP have their own advantages and disadvantages. Who can become the spiritual leader of the next-generation network?
H.323: Mature but not all-encompassing
H.323 was created in 1996 and an upgraded version 2 was introduced in January 1998. H.323 is said to be an all-encompassing standard because it itself consists of numerous subordinate protocols. Because of this, the ITU can define H.323 by using many existing data and communication standards such as Q.931, G.711, and G.723.1.
H.323 was initially proposed by Intel and PictureTel. This protocol defines a communication method that can be flexibly applied to multimedia conferencing equipment and provides application sharing features on the IP stack. Designers have proposed standards applicable to a variety of devices, including video phones, desktop computers, and large multi-port gateways. Therefore, H.323 is extensive and provides multiple media types and compression techniques applicable to different devices.
The core advantage of H.323 lies in its maturity, which helps many software vendors develop stable devices and also helps different vendors eliminate problems in interoperability and launch various devices supporting the H.323 standard in the market. Because the H.323 standard incorporates the Q.931 call control protocol, many developers with rich experience in existing ISDN telephone technology are also very familiar with this call control model. In fact, events and parameters can often directly enter the application system that previously worked under ISDN through H.323. When defining H.323, the designers started from the perspective of terminal devices rather than the internal devices of the existing PSTN. Therefore, H.323 cannot be integrated with SS7 or supplement the powerful functions that SS7 must provide. In addition, the scalability of H.323 has been proven to be a problem in very large-scale applications. When designers use gateways with tens of thousands of ports, they find that centralized state management is a bottleneck.
The market's response to H.323 indicates that the best application point for H.323 should be systems located at or near the endpoints with 1 to 200 ports. H.323 works well in an environment with sufficient processing power to implement call control and media processing. H.323 has received the strongest support from the industry as an enterprise IP telephony solution.
MGCP: Excellent Interoperability
The Media Gateway Control Protocol (MGCP) provides a solution for the interconnection of numerous IP telephony gateways, enabling these gateways to be interconnected into an interoperable entity. MGCP assumes that the call agent (CA) performs all call control processing, while the media gateway controller (MGC) performs all media processing and conversion.
The specification of MGCP was developed by different companies (such as Telecordia and Lucent) and was published by the IETF in the form of an RFC (RFC 2705) information document. MGCP is the result of the combination of the Simple Gateway Control Protocol (SGCP) and the IP Device Control (IPDC) protocol, but it has not yet been recognized. The Megaco Working Group of the IETF, in collaboration with ITUA, is working on developing the recommended standard H.248 based on MGCP (formerly H.gcp). The core document and related specifications were completed in February 2000 and were published as an IETF RFC standard document
When H.323 gateways provide media conversion and SS7 gateways translate call control information, MGCP can be used in conjunction with H.323 gateways and SS7 gateways. In this case, MGCP transmits all call control information from endpoint devices to the network. Using this structural approach, developers can supplement the capabilities of the SS7 network and build larger IP telephony systems compared to using H.323 alone.
To coordinate the media path and performance of a single call, MGCP relies on the Session Description Protocol (SDP), which is part of the MGCP specification. SDP allows for negotiation on parameters such as RTP ports and endpoint IP addresses, voice coding methods (such as G.711 and G.723.1), packet periods, and other connection type parameters.
The advantages of MGCP include: It is particularly suitable for configuring large application systems because it is itself used to address specific issues of large systems. Using MGCP can achieve good integration with the SS7 network and provide greater control and throughput for call processing. MGCP separates media processing and signaling functions, enabling the development of simpler systems by multiple equipment providers.
Some of the shortcomings of the MGCP protocol include: MGCP is overly complex for small application systems. MGCP is in competition with the H.248/Megaco standard, which was signed and recognized by the IETF and ITU in early 2000. Thus, operators requiring MGC can choose either MGCP or H.248. Therefore, H.248 may eventually replace earlier versions of MGCP. The destination of MGCP is the telecommunications operation market, where it enables the transmission of tens of thousands of IP telephony calls.
SIP: The New Generation Pushes the Old
The Session Initiation Protocol provides a method for transmitting call control information either between terminal devices or proxy servers, or to gateway devices. This is the result of the efforts of the IETF MMUSIC Working Group. Similarly, like many existing Internet protocols, SIP also contains the commonly used HTTP protocol.
SIP is considered a lightweight protocol because it uses simple text commands that can be easily generated and analyzed by terminal devices. SIP only uses 6 instructions to manage call control information. The simplicity and ease of use of the SIP protocol is an important reason for very low-cost application systems to choose this protocol.
SIP does not define the media transmission mechanism, so this protocol can be used in application systems where media transmission is a dedicated device, which can improve efficiency and reduce costs. SIP also allows call control information to be transmitted through any datagram protocol, making it effective in non-TCP/IP environments (such as Novell or other proprietary protocols).
Some of the advantages of SIP include: The protocol has scalability and new functions can be easily defined and quickly implemented. It can be easily embedded in inexpensive end-user devices. The protocol ensures interoperability and enables different devices to communicate. It is easy for developers outside the telephone field to understand the protocol.
The disadvantages of SIP include: SIP has just emerged, so most applications are still in the prototype stage. The application scope of this protocol alone is relatively narrow. However, when used in conjunction with other protocols, it has strong flexibility. SIP is only a small part of a complete solution, and many other software are needed to build a complete IP telephony product.
Low-cost terminal products are undoubtedly the most natural application of SIP. Devices such as wireless phones, top-mounted splitters, Ethernet phones, and other devices with limited computing and memory resources can use this protocol. Since SIP is a superior call control protocol, it is currently the first choice to replace the MGCP call control protocol.
Each of the above protocols emphasizes different aspects of the technology required for developing IP telephony systems. Many of the systems currently under development contain at least one of these protocols, and these protocols often require interoperability. All these protocols continue to evolve in the process of building complex IP telephony systems. Manufacturers are all attempting to develop interoperable systems, so interoperability will continue to be a major challenge in the future. The new standard protocol MGC (H.248/Megaco) derived from the IETF and ITU is expected to become the strongest competitor to MGCP in the transmission market.
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SIP Has Become the Trend
In recent years, a very obvious change has occurred in the process of building the next-generation network: Many equipment manufacturers have begun to tilt the protocol standard of their next-generation networks towards SIP, including Cisco and Nortel, which have achieved significant improvements many years ago.
SIP has become the pursued standard, and the biggest advantage is its simplicity in development compared to the previous H.323 standard. When IP and multimedia have become an irreversible trend, it is reasonable that SIP is sought after. At the same time, SIP is inheritable. For operators with traditional networks, SIP is undoubtedly the best entry point for network development.
Today, when users have diverse requirements for network service quality, the original H.323 protocol has been unable to provide a higher level of service quality guarantee, while the SIP protocol ensures good network service quality. With the acceleration of productization, many manufacturers have completed the productization of solutions based on the SIP protocol, and this has become a key indicator requirement before network construction tenders by operators.
The use of the SIP standard in 3GPP to support voice and data is an important reason for the development of the SIP protocol. SIP can optimize voice well, and due to its programmability, it provides a good guarantee when mobile services face changes in flexibility and diversity.
In addition, the rich and colorful terminals based on SIP are very beneficial to the completion of the complete industrial chain from the network to access to the end user. This is what operators lack the most today in the absence of balanced development factors in the industrial chain. (Zhang Jie, Researcher at the Telecommunication Research Institute of the Ministry of Information Industry)
What Does the Next-Generation Network Need? SIP: The Simpler, the More Popular.
Almost at the same time, various forces in telecommunications and the Internet have begun to favor SIP. First, 3GPP adopted SIP as the core protocol of the all-IP network for the third-generation mobile communication, which made all 3G equipment manufacturers start efforts to comply with the SIP protocol to achieve interconnection; then, the protocol of the NetMeeting component in the Windows XP operating system was also changed from H.323 to SIP protocol, which means that even the most valuable software terminal for instant messaging has also started to follow the SIP path. For both the telecommunications network and the Internet, in order to achieve flexibility in business operations, the SIP protocol has become the direction of future network development.
SIP is a telephone signaling protocol that comes from the traditional telecommunications camp, so it inherently brings the characteristic of having a complex architecture. It can provide good transparency for implementing services in the existing circuit-switched telephone network in the softswitch. SIP is mainly used to support multimedia and other new services, and has more flexible and convenient characteristics in multi-service applications based on IP networks.
However, SIP is not perfect. Relatively speaking, SIP is not as mature as BICC in voice services, but it can support stronger multimedia services and has good scalability. It can be extended accordingly based on different applications. In the application of softswitch in the fixed network, the SIP protocol is in the call control layer of the flat architecture and provides support for call connection between different softswitches. When adopting the SIP architecture, from the perspective of routing, there are two situations: The first situation is that normal ISUP messages are encapsulated in SIP messages after adding some information, and the functions such as call server, number, routing analysis and signaling, and service interworking remain unchanged, and the routing analysis guides the addressing to the target IP address. The second situation is based on the ENUM database. In this way, the call control of the call server is completely different from the call control in the existing circuit-switched network. There will be no numbers and routing analysis in the call control, but service mapping and interworking are still required.
In addition, compared to the existing network, operators have less control over the network, and the control method has undergone a huge change. If some functions are to be introduced, the SIP protocol needs to be extended.
The SIP-I (SIP with Encapsulated ISUP) protocol series includes TRQ.BICC/ISCUPSIP and Q.1912.SIP of the ITU-T SG11 Working Group. The former defines the technical requirements for the interworking of SIP and BICC/ISUP, including the interworking interface model, the protocol capability set that the interworking unit IWU should support, the security model of the interworking interface, etc. The latter, based on the protocol capability configuration sets A, B, and C that the IWU should support on the NNI on the SIP side of 3GPPSIP, defines in detail the interworking of 3GPPSIP and BICC/ISUP, the interworking of SIP and BICC/ISUP in general situations, etc.
Most importantly, the SIP-I protocol series has the inherent clarity, accuracy, and detail of ITU-T standards, is highly operable, and 3GPP has adopted Q.1912.SIP as the final standard for the interworking of 3GPPIMSR5 with PSTN/PLMN. Therefore, the standardization of the interworking of China's NGNSIP with PSTN/ISDN should be based on the ITU-T SIP-I protocol series. In fact, major Chinese telecom operators have ultimately chosen SIP-I and abandoned SIP-T.
According to experts, the application of SIP in the NGN environment needs to meet many new requirements of the telecommunications network. The standardization within the NGN SIP domain should mainly be carried out in aspects such as network architecture, operator's control of calls and sessions, billing, security, QoS, routing, and service implementation, and the standardization of NGN SIP should be completed on the basis of IETF SIP. Because the ITU-T SIP-I protocol series is more comprehensive, clear, accurate, and operable than the IETF SIP-T protocol series, the interworking of NGN SIP networks with traditional PSTN/ISDN should adopt the ITU-T SIP-I protocol series.
Related Information
SIP (Session Initiation Protocol) is an IP telephony signaling protocol proposed by the IETF (Internet Engineering Task Force). Based on the SIP protocol standard, it integrates traditional voice and value-added services, provides the latest instant messaging services and video services on the IP network, and can provide a standard and highly scalable platform for many other value-added application service providers. The system platform fully adopts the distributed architecture of the Internet, has high flexibility and scalability, and has the high reliability and fault tolerance required by large-scale telecommunications services, and can support millions of users.
