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What are the differences between H.323 and SIP?

Latest reply: Apr 15, 2022 08:43:36 6326 24 19 0 0

This post is about the differences between H.323 and SIP. Please see below for details.


H.323 and SIP are protocols used in the communications field and the Internet domain respectively. Both protocols have requirements for voice coding and decompression. They provide a complete solution to the (VoIP) signaling of the IP network telephone system. They have the capability of establishing, managing, and releasing call connections, and have the functions of network management and capability exchange, the call setup and interaction of end users have the QoS capability, and the new functions are easy to be expanded to support different types of interoperation.


H.323 and SIP have a lot in common and have certain overlap in positioning. In addition, H.323 and SIP have become the two competing protocols in the packet network because of the development of the protocol and the rapid expansion of the network. What are the differences between H.323 and SIP?


H.323 and SIP


What is H.323

H.323 is an agreement proposed by ITU-T Working Group 16, which is a set of agreements. There are signaling for encoding, decoding, and encapsulation of voice and video signals, signaling for sending and receiving call signaling, and signaling for capability exchange. The H.323 protocol family defines the components, protocols, and procedures for providing multimedia communication on a packet switched network PBN (such as an IP network).


H.323 supports point-to-point or point-to-multipoint communication of audio, video, and data. The H.323 protocol family includes an H.225.0 for establishing a call, an H.245 for control, an H.332 for a large conference, an H.450.X for supplementary services, a secure H.235, and an H.246 for interworking with a circuit switched service.

H.323 protocol stack


What is SIP

SIP (Session Initiation Protocol) is an applicatication control (signaling) protocol implemented by the IETF MUSIC team in 1999 to implement real-time communication. It can be used to create, modify, and terminate multimedia session processes involving multiple participants. The (Session) refers to the data exchange between users.


SIP handshake


The members participating in the session can communicate in multicast mode, unicast network mode, or combination mode. It is a part of the Internet multimedia communication and control protocol system, including the Session Description Protocol (SDP), Session Release Protocol (SAP), and Session Initiation Protocol (SIP). Such sessions can be voice, video, text chat, interactive games, or even virtual reality.



The basic structure of the SIP network consists of the user agent and IP network. The IP network includes various network servers required by the SIP system. The user agent has the user agent client (UAC) and the user proxy server (UAS). There are also two types of network servers, which are proxy servers and redirect servers.


What are the differences?



From the perspective of origin:


The fundamental power of the H.323 protocol is the integration of three networks. At that time, the IP technology and Ethernet technology developed rapidly. Many enterprises and enterprises had their own LANs. Therefore, a network with voice, video, and data services based on the Ethernet or IP network was urgently required. Therefore, the H.323 protocol came into being, it is an agreement in the communications field. The SIP protocol is a multimedia communication protocol based on computer and network proposed by computer workers in the case of rapid Internet development.

It is a protocol in the field of computer network. Because of different starting points, their control structure is also different. In terms of call control and signaling, H.323 mainly refers to the call control and signaling architecture of the traditional PSTN (public switched telephone network). The PSTN is a

hierarchical, master-slave, and centralized control mode for call control and signaling. The H.323 also uses this control mode. The Internet is a distributed, client/server, and horizontal control network. Therefore, the communication mode implemented by the SIP protocol is based on a distributed, client/server, and horizontal control structure.


Different design ideas, reference different standards and different development purposes, make the two major agreements distinctive. The H.323 protocol uses the ISDN design idea. It uses the Q.931 protocol to establish and release calls. It has the working mode of centralized processing and management of the telecommunication network, and has the capability of making an IP phone system of any scale. The H.323 development aims to provide users with VoIP and video communication services that replace common phones on the packet switched network.


As an IETF standard, SIP greatly draws on other widely used Internet protocols, such as Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP). It is also a text protocol.

Based on the experience of successful web services, the SIP protocol is used to construct the IP phone service network based on the existing Internet.


In terms of time:


H.323 is earlier than SIP. It is more strict than SIP in terms of management control and QoS mechanism, and is more complex.

The Internet Engineering Task Force (IETF) has formulated the SIP protocol based on the H.323. Compared with the H.323, the SIP is simpler, better expandable, and more closely related to the existing Internet applications.


The H.323 uses the traditional mode of implementing the telephone signaling, which complies with the traditional design idea in the communication field and implements centralized and hierarchical control. The H.323 protocol is used to connect to the traditional telephone network. Therefore, the related products are more abundant. Due to the short launch time, the SIP protocol is not as mature as the H.323 protocol. It is inconvenient to connect to many existing traditional networks. However, due to the excellent connectivity with the Internet, more and more emerging network products support SIP, which presents a broad development prospect.


The key to implementing IP telephony is the signaling protocol. According to the signaling protocol, the IP phone supported by the H.323 is not much different from the traditional telephone on the surface. The difference is that the circuit switching mode of the traditional telephone is changed to the current packet switching mode. SIP is an application that expands IP phones into the Internet. Compared with other network applications such as FTP and E-mail, SIP adds signaling and QoS requirements. From this perspective, both of them support the IP phone service and use the RTP (real-time transport) as the media transmission protocol.


From the coding method of the message:


The H.323 protocol supports the ITU-T standard and uses the binary method based on the ASN.1 and compression coding rules to represent the message. ASN.1 usually requires a special code generator for analysis, which is complex.


SIP has the capability to support any codec protocol. Because SIP is based on text, its code generation and syntax parsing are simple. Sip coding means that the meaning of the header is clear, such as From, To, and Subject, moreover, it also facilitates the extension and debugging of the protocol, improves the degree of user localization, and has fully embodied its superiority in practice.


The scope of use of H.323 and SIP


Currently, many countries, including China, use H.323 as the protocol between IP phone gateways. The entire IP phone system uses only the IP network as the transmission medium. The IP phone gateway is used as the interface between the circuit switched network and the IP network.

The H.323 adopts strict centralized control. The reliability is high and the call processing capability is reduced. This shows that the reliability of the H.323 is more flexible than that of the SIP. Therefore, SIP is usually used in mass instant messaging, such as individuals and enterprises, that have low reliability requirements. In the industry network, for example, for videoconferencing, H.323 is used.


In addition, the H.323 protocol is widely used in professional videoconferencing. This is because the videoconferencing technology is mainly used for videoconferencing service operation or internal videoconferencing. The network structure is hierarchical and domain-based.


H.323 is especially suitable for this structure, and H.323 is centralized control and facilitates charging, broadband management is simple and effective.

However, the H.323 is not perfect in application. The H.323 does not support the multicast function of the signaling. The function of the H.323 does not support the scalability and reduces the reliability. Compared with SIP, SIP is a distributed call model and has a distributed multicast function. The multicast function facilitates conference control, simplifies user positioning and group invitation, and saves broadband. In addition, SIP has a strong development space for conference calls, if a vendor uses a computer to implement the videoconferencing system, the SIP protocol is often used to implement the system. The SIP has received many responses, including communications device manufacturers and Internet service providers. They increasingly claim that their devices support SIP.


Can H.323-registered endpoints communicate with SIP-registered endpoints?


No. 

The two protocols cannot communicate with each other. However, you can use dual registration. The endpoint registers with both H.323 and SIP. In this way, a dual-registration terminal can communicate with both the H.323-registered terminal and the SIP-registered terminal.

Let’s Sum Up


Although the H.323 protocol and SIP protocol propose two sets of IP phone system structures, they pursue the same objectives. They are completely independent of each other and cannot be compatible with each other. That is, only one of them can be selected in one implementation. Although the SIP protocol is later than the H.323 protocol and is formulated after analyzing many H.323 problems, SIP is not an upgrade version of the H.323. They are developed and modified separately based on different development application systems.


However, H.323 and SIP can communicate with each other. Different protocols can be used in different phases to achieve the best solution. From the development history of H.323 and SIP, they play an important role in the application of VoIP. Looking into the future NGN, both will play an important role.


Related topic:

Difference between x264 and h264.

H.323

SIP


Architecture

H.323 covers almost every service, such as capability exchange, conference control, basic signaling, QoS, registration, service discovery, and so on.

SIP is modular because it covers basic call signaling, user location, and registration.  Other features are in other separate orthogonal protocols.

Components

Terminal/Gateway

UA

Gatekeeper

Servers

Protocols

RAS/Q.931

SIP

H.245

SDP

Call control Functionality

Call Transfer

Yes

Yes

Call Forwarding

Yes

Yes

Call Holding

Yes

Yes

Call Parking/Pickup

Yes

Yes

Call Waiting

Yes

Yes

Message Waiting Indication

Yes

No

Name Identification

Yes

No

Call Completion on Busy Subscriber

Yes

Yes

Call Offer

Yes

No

Call Intrusion

Yes

No


H.323 splits them across H.450, RAS, H.245 and Q.931


Advanced Features

Multicast Signaling

Yes, location requests (LRQ) and auto gatekeeper discovery (GRQ).

Yes, e.g., through group INVITEs.

Third-party Call Control

Yes, through third-party pause and re-routing which is defined within H.323. More sophisticated control is defined by the related H.450.x series of standards.

Yes, through SIP as described in separate Internet Drafts.

Conference

Yes

Yes

Click for Dial

Yes

Yes

Scalability

Large Number of Domains

The initial intent of H.323 was for the support of LANs, so it was not inherently designed for wide area addressing. The concept of a zone was added to accommodate wide area addressing.  Procedures are defined for “user location” across zones for email names. Annex G defines communication between administrative domains, describing methods to allow for address resolution, access authorization and usage reporting between administrative domains. In multi-domain searches, there is no easy way to perform loop detection. Performing the loop detection can be done (using the PathValue field), but introduces other issues related to scalability.

SIP inherently supports wide area addressing. When multiple servers are involved in setting up a call, SIP uses a loop detection algorithm similar to the one used in BGP, which can be done in a stateless manner, thus avoiding scalability issues. The SIP Registrar and redirect servers were designed to support user location.

Large Number of Calls

H.323 call control can be implemented in a stateless manner.  A gateway can use messages defined in H.225 to assist the gatekeeper in performing load balancing across gateways.

Call control can be implemented in a call stateless manner. SIP supports n to n scaling between UAs and servers. SIP takes less CPU cycles to generate signaling messages; therefore a server could theoretically handle more transactions. SIP has specified a method of load balancing based upon the DNS SRV record translation mechanisms.

Connection State

Stateful or Stateless.

Stateful or Stateless.  A SIP call is independent of the existence of a transport-layer connection, but instead signals call termination explicitly.

Internationalization

Yes, H.323 uses Unicode (BMPString within ASN.1) for some textual information (h323-id), but generally has few textual parameters.

Yes, SIP uses Unicode (ISO 10646-1), encoded as UTF-8, for all text strings, affording full character set neutrality for names, messages and parameters. SIP provides for the indication of languages and language preferences.

Security

Defines security mechanisms and negotiation facilities via H.235, can also use SSL for transport-layer security.

SIP supports caller and callee authentication via HTTP mechanisms. Cryptographically secure authentication and encryption is supported hop-by-hop via SSL/TSL, but SIP could use any transport-layer or HTTP-like security mechanism, such as SSH or S-HTTP. Keys for media encryption are conveyed using SDP. SSL supports symmetric and asymmetric authentication. SIP also defines end-to-end authentication and encryption using either PGP or S/MIME.

Interoperability among Versions

The fully backward compatibility in H.323 enables all implementations based on different H.323 versions to be seamlessly integrated.

In SIP, a newer version may discard some old features that are not expected to be implemented any more. This approach saves code size and reduces protocol complexity, but loses some compatibility between different versions.

Implementation Interoperability

H.323 provides an implementers’ guide, which clarifies the standard and helps towards interoperability among different implementations.

SIP thus far has not provided an implementation agreement.

Billing

Even with H.323's direct call model, the ability to successfully bill for the call is not lost because the endpoint reports to the gatekeeper the beginning and end time of the call via the RAS protocol.

If the SIP proxy wants to collect billing information, it has no choice but to stay in the call signaling path for the entire duration of the call so that it can detect when the call completes. Even then, the statistics are skewed because the call signaling may have been delayed.

Codecs

H.323 supports any codec, standardized or proprietary, not just ITU-T codecs. There have been codepoints for MPEG and GSM, which are not ITU-T codecs, in H.323 for a long time; many vendors support proprietary codecs through ASN.1 NonStandardParameters, which is equivalent to SIP's "privately-named codec by mutual agreement"; and any codec can be signaled via the GenericCapability feature that was added in H.323v3. Payload types can be specified statically or dynamically.

SIP supports any IANA-registered codec (as a legacy feature) or other codec whose name is mutually agreed upon. Payload types can be specified statically or dynamically.

Call Forking

H.323 gatekeeper can control the call signaling and may fork the call to any number of devices simultaneously.

SIP proxies can control the call signaling and may fork the call to any number of devices simultaneously.

Transport protocol

Reliable or unreliable, e.g., TCP or UDP. Most H.323 entities use a reliable transport for signaling.

Reliable or unreliable, e.g., TCP or UDP. Most SIP entities use an unreliable transport for signaling.

Message Encoding

H.323 encodes messages in a compact binary format that is suitable for narrowband and broadband connections.

SIP messages are encoded in ASCII text format, suitable for humans to read.

Addressing

Flexible addressing mechanisms, including URLs and E.164 numbers.

SIP only understands URL-style addresses.

PSTN Interworking

H.323 borrows from traditional PSTN protocols, e.g., Q.931, and is therefore well suited for PSTN integration. However, H.323 does not employ the PSTN's circuit-switched technology--like SIP, H.323 is completely packet-switched. How Media Gateway Controllers fit into the overall H.323 architecture is well-defined within the standard.

SIP has no commonality with the PSTN and such signaling must be "shoe-horned" into SIP. SIP has no architecture that describes the decomposition of the gateway into the Media Gateway Controller and the Media Gateways.

Loop Detection

Yes, routing gatekeepers can detect loops by looking at the CallIdentifier and destinationAddress fields in call-processing messages. If the combination of these matches an existing call, it is a loop.

Yes, the SIP message Via header facilitates this. However, there has been talk about deprecating Via as a means of loop detection due to its complexity. Instead, the Max-Forwards header seems to be the prefered method of limiting hops and therefore loops.

Minimum Ports for VoIP Call

5 (Call signaling, 2 RTP, and 2 RTCP.)

5 (Call signaling, 2 RTP, and 2 RTCP.)

Video and Data Conferencing

H.323 fully supports video and data conferencing. Procedures are in place to provide control for the conference as well as lip synchronization of audio and video streams.

SIP has limited support for video and no support for data conferencing protocols like T.120. SIP has no protocol to control the conference and there is no mechanism within SIP for lip synchronization.

Microtronix Test System Available

Yes

Yes


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https://forum.huawei.com/enterprise/en/what-is-the-difference-between-x264-and-h264/thread/700917-885

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