H323: protocol, gateways, ports, configuration and application

The H323 protocol standard is gradually out of circulation, in particular for telephones. A few years ago, many phone manufacturers abandoned the trend of creating devices that could use both IP-telephony protocols. Today, SIP is mainly used. Nevertheless, many users have a H323 device and they need to know the rules of operation and settings. The standard still offers better network management and high-quality hardware compatibility. The difference between the two protocols is decreasing with each new version, despite numerous debates on this issue in the industry.

Briefly about IP-telephony

Video over the Internet is a technology that allows you to transfer voice and data on the same network based on the IP protocol. The term converged networks or IP convergence is often used, which implies a broader concept of integration of all communications: voice, data and video.

This technology has been on the market since the late 1990s, but only recently has it become widespread due to the improvement and standardization of speech quality control (QoS) systems and the universalization of the Internet service.

IP-telephony system - a set of elements that are properly integrated and use services based on VoIP companies. The main elements of this system are: IP-PBX, IP-gateway and various IP-telephones with H323 protocol.

Business Communications Convergence - Sound, Data, and Video. The IP network is a modern trend and gives users important advantages:

1. Save on calls.
2. Simplification of communication infrastructure.
3. Management optimization.
4. Unification of a telephony system between objects.
5. Mobility / accessibility for the user.

A multiservice converged network must be properly designed and managed, and aspects such as reliability, security and quality of service (QoS) control are needed to ensure an optimally functioning system.

Two major video calling standards

SIP and H323 are considered the main video conferencing standards. Different organizations are considering transmitting audio and video signals over IP with different approaches. The International Telecommunication Union (ITU) has established H323 as the first IP multimedia communication protocol. SIP is an Internet Engineering Task Force (IETF) approach for transmitting audio and video over IP.

The H323 protocol is an umbrella format that provides a well-defined system architecture, including call control and multimedia. While the H323 uses a telecommunications approach for voice / video over IP, SIP uses an Internet approach.

Network entities: gateways, terminals, and conference bridges, as well as the gatekeeper. The architecture is peer-to-peer and supports communication between users without a centralized monitoring facility. H323 call information is recorded in binary code with a specific set of translations for each code. This was done to reduce transmission size and save bandwidth. New codes must have an agreed definition between the parties before the call. The standard can be updated, but any additions to the standard require backward compatibility with the existing one.

Operational Protocol H323

The International Telecommunication Union has defined the H323 standard for providing audiovisual communication sessions over network packets for voice over IP and video conferencing on the same basis. It transmits messages, offers directory services, terminal access control, resource consumption control and processes call authorization, and can also route signaling. The standard has access to other networks, performing the functions of data conversion and broadcast signals.

Main characteristics:

1. Guaranteed Quality of Service (QoS).
2. It does not depend on the network topology.
3. It supports gateways and uses at the same time more than one channel: voice, video, data.
4. Allows companies to add functionality to implement the necessary interaction functions.

The main components of the system

H323 implements the following basic components: terminals, gateways for connecting to PSTN / IN resources, gatekeepers for access control, registration and bandwidth, MCU (multiconference) and control units. Description of the H323 protocol and its components:

1. Terminals - equipment used by users, can be implemented either programmatically using a computer, or hardware - physically.
2. Guardians (GateKeepers) - are the center of each VoIP-organization and the equivalent of private branches or UATS (Private Branch eXchange). They usually come up with software.
3. Gateways - connected by switching to a public telephone network, acting transparently to the user.
4. Multipoint control units - are responsible for managing the conference.

A generally accepted standard has been distributed across the wide area network (WAN). It consists of many terminals that it manages. For example, several local networks separated by routers.

The H323 protocol stack runs on top of the transport and network layers. If the core network is IP-based, then the audio, video, and H.225.0 RAS packets use the UDP (H.245) protocol for data transfer and control. H.225.0 packet call signaling is transported using the reliable TCP transmission control protocol.

Protocol Technologies

An example of enterprises using H323 is an IP-telephony company that implements Asterisk PBXs with the cost of billing telephone calls to landlines and mobile phones. The user can transfer to a mobile phone, transfer the voice to the phone, make calls, record calls, and report PBX calls. Asterisk can be implemented on the Linux platform, so there will be no software licensing costs. Benefits of VoIP PBX:

1. Supports SIP, IAX, H323, MGCP, SKINNY and others.
2. Supported codecs: ADPCM, G.711 (A-Law & -Law), G.722, G.723.1 (pass-through), G.726, G .729 (if purchased a commercial license), GSM, iLBC.

IP ports and protocols used by various H323 device providers.

Message exchange

The H323 protocol exchanges messages between the two terminals and establishes who will be the master and who will be the slave, as well as the bandwidth of the participants and the audio and video codecs that will be used. The terminals initiate the exchange of data, audio or video via the RTP / RTCP protocol. During the disconnect phase, any active participant in the communication can initiate the call termination process using the Close Logical Channel and End Session Command messages, after which the connection to RELEASE is closed.

The standard was originally developed to provide audio, video, and shared communications between hosts connected to a corporate LAN and remote devices of a traditional PSTN circuit-switched network. It implements the H323 SIP gateway controller, which is responsible for authentication and location of users, tracking registered clients.

In the process, the proxy gatekeeper communicates with the client, which reduces the load on low-power client devices. Each terminal is identified by a pair (IP address, TCP / UDP port), so you can directly contact it through its address / port pair without using a gatekeeper. If there is a gatekeeper, address / port pairs can be mapped to aliases so that users can remember them better, for example, name@domain.com. Since they are associated with user accounts, aliases allow nomadism - the user will remain available even if he moves by changing his IP address.

H323 Call Stages

The call takes place in several main stages:

1. Registration - calls the terminal, searches for the gatekeeper in its zone and opens the RAS channel using its control.
2. Call setup — The caller’s terminal sets the channel for the called party’s terminal using call control.
3. Negotiation - parameters, such as bandwidth and codecs, are negotiated using the control.
4. Data transmission - voice is transmitted via RTP closing, the data channel is closed using the control.
5. Termination - The RAS channel is closed using the control.

The gatekeeper can play two roles:

1. A routed call passes through the gatekeeper, which is useful for bypassing NAT, so the gatekeeper acts as a relay server.
2. Gatekeeper Direct Endpoint - The call is routed directly to the endpoint, but first, the calling and called party clients must complete the Admission step to charge and manage bandwidth.

Global Protocol and Modification Differences

The Session Initiation Protocol (SIP) standard developed by the MMUSIC IETF working group is designed to initiate, modify, and terminate user sessions: video, voice, and quick messaging.

The syntax of its operations resembles the syntax of HTTP and SMTP, the protocols used in web page services and e-mail distribution. This similarity is natural because SIP was designed to make telephony another service on the Internet. This is a new standard for establishing, routing, and modifying communication sessions over Internet Protocol (IP) networks. He uses the Internet model and transforms it into the world of telecommunications, using existing Internet protocols such as HTTP and SMTP (Simple Mail Transfer Protocol).

It also uses a URL structure to identify users instead of devices. Thus, SIP is device independent and does not distinguish between voice and data, phone or PC. SIP is more used for service management, while the H323 performs the function of converting the telephone standard into IP packets.

The H323 was introduced as an evolution of SS7 designed to control circuit-switched signaling. On the contrary, SIP is closer to HTTP, the packet network paradigm used on the Internet. Looking ahead, it is better to choose SIP. In this case, streams of multimedia information are transmitted using RTP, so the choice of a control protocol or another does not directly affect the quality of the services offered. H 323 is much more complicated than SIP. It has hundreds of different messages encoded in binary format. Therefore, the H323 provides the work of developers as well as network administrators in troubleshooting.

Interoperability Scenarios

These protocols are widely used, so interoperability between SIP and H323 is necessary to ensure full end-to-end connectivity. Due to the inherent differences between H323 and SIP, it is necessary to ensure consistency for interoperability. If they are used in the same administrative domain, call setup messages must be translated, and then RTP can be used for communication between the SIP telephone and the H323 telephone.

The scenario becomes more complex when the IP and the H 323 gateway operate in separate administrative domains. This needs to be interpreted using a different protocol. H323 defines conference calling as part of the standard, including centralized and decentralized. SIP does not have a definition for conferencing, but there is a process for conferencing that is similar to H323 but has not been formally defined as part of the standard.

Working with the Cisco C90 Codec

Networking H323:

1. First, codecs are configured taking into account IP addresses (192.168.2.XXX), the correct subnet mask (255.255.255.0), the gateway (192.168.XXX) - the pfSense IP interface and the DNS server. On the C90, this is in the administrator settings -> IP settings.
2. The NAT version for H323 is located in Admin Settings -> Advanced -> H323 -> NAT.
3. Set: Mode = On.
4. Make sure the NAT address = WAN address so that the code can send the correct packets to the original connection.
5. Set the codec command to ban connection through the gatekeeper. Under administrator settings -> Advanced -> H323 -> Profile 1 -> CallSetup.
6. Set: Mode = Direct.
7. H323 uses static ports for connections. Admin Settings -> Advanced -> H323-> Profile 1.
8. Set: Port Allocation = Static.
9. Recheck the H323 IP setting -> Advanced -> Network 1 and make sure that the IP settings match the form.
10. Check the RTP ports to be used. The RTP stream transmits and receives audio and video. Under administrator settings -> Advanced -> RTP -> Ports -> Range. Pay attention to the Start and Stop values, you need to add them to the port firewall forwarding rules. The default is usually 2326/2486.
11. Restart the codec for the settings to take effect.

After adding these rules, you can receive an H323 call from the codec. If calls do not go through, you need to look in the logs to see if certain ports are being requested. You might need to configure your H323 port forwarding settings and rules.

Benefits of Choosing IP Telephony

With VoIP, you can call from any direction where there is a network connection. Since IP phones transmit information over the network, they are controlled by providers from anywhere in the connection. This is an advantage for people who usually travel a lot and can carry their phone with them, having access to the IP-telephony service.

The benefits of VOIP services include:

1. Call identification.
2. Call Waiting Service.
3. Call forwarding service.
4. The second call.
5. Back call.
6. Call 3 lines (three way call).
7. Call forwarding to a specific phone.

Obviously, both voice and video communication standards have their drawbacks and advantages, therefore, instead of focusing on one standard compared to another, it will be more rational to work on improving ways to ensure interoperability between the standards. This will allow for end-to-end connections throughout the network, and provide additional IP-oriented services. It is this modern approach that will demonstrate the power of IP communications.