5. The H.

323 Standard

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5.1 The H.323 Protocol Suite (1/2) .....................................................3

5.1 The H.323 Protocol Suite (2/2) .....................................................4
5.1.1 H.225 RAS Control.....................................................................5
5.1.2 H.225 Q.931 Call Signaling........................................................6
5.1.3 H.245 Call Control Signaling.....................................................7
5.1.4 Audio and Video Codecs (1/2) ..................................................8
5.1.4 Audio and Video Codecs (2/2) ..................................................9
5.1.5 T.120 ......................................................................................... 10
5.1.6 The RTP Protocol .................................................................... 11
5.1.7 H.235 Security.......................................................................... 12
5.1.8 H.450 Zusatzdienste ................................................................ 13
5.1.9 H.450 Supplementary Services .............................................. 14
5.2 H.323 over IP based Networks (1/4) .......................................... 15
5.2 H.323 over IP based Networks (2/4) .......................................... 16
5.2 H.323 over IP based Networks (3/4) .......................................... 17
5.2 H.323 over IP based Networks (4/4) .......................................... 18

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5.1 The H.323 Protocol Suite (1/2)

The H.323 standard, version 1, was finaliezed by the ITU in May 1996. At that time H.323
was focused on multimedia communication within a LAN. In version 2 from January 1998
and version 3 from September 1999 the H.323 protocol described data communication in
audio, video and data conferences via networks without guaranteed quality of service (QoS).
Thus H.323 is also valid for IP networks like the Internet.
H.323 is an independent standard ensuring the compatibility between products of different
manufacturers. Further more interoperability with existing circuit switched networks is
supported.
The ITU H.32x protocol family deals with the transmission of multimedia data over different
types of connections. H.323 is quite similar to H.320 which handles ISDN communication
with guaranteed QoS, e.g. videoconferencing. The IMTC (International Multimedia
Teleconferencing Consortium) sees H.323 as an extension to H.320 for intranet and packet
switched networks. H.323 is completely independent from the transport network. The
standard makes no provission for ensuring a dedicated QoS.

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5.1 The H.323 Protocol Suite (2/2)

Before a call can be established between two terminals a lot of control information has to be
exchanged in addition to audio and video data. The required protocols are specified in ITU
H.225.
H.225 includes call signaling, i.e. Q.931 with call setup and release messages, and RAS
(Registration Admission Status) signaling for communication to the Gatekeeper. H.245 Is
responsible for call control, the exchange of information concerning the capability of the
terminals and the setup of a logical channel for real time data transmission. Together, H.225
and H.245 make up the system control unit. A call is a point-to-point communication between
two H.323 endpoints. These could be terminals, gateways or Multipoint Control Units
(MCUs).

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5.1.1 H.225 RAS Control

Registration Admission Status (RAS) is the protocol between a terminal and the Gatekeeper
of the H.323 network. It is used for signaling, e.g. registration at the gatekeeper, checking
permission before using a service and status information exchange between terminal and
gatekeeper.
Registration means that each terminal is recorded by a central entity in the H.323 network.
This is necessary for address translation from E.164 or alias names into IP addresses as
performed by the gatekeeper. Additionally it make sure that only a registered terminal may
receive a call.
The permission includes a call request and acknowledgement cycle between a terminal and
the gatekeeper. Each terminal action has to be requested at the gatekeeper which accepts or
rejects the request. With this mechanism it is possible to create individual access profiles for
each user, e.g. a user is might be allowed to perform national calls only. The transmission of
RAS messages is handled on an RAS channel set up between the terminal and the
gatekeeper prior to seting up any other channel, e.g. call setup towards another terminal.

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5.1.2 H.225 Q.931 Call Signaling

Q.931 handles call setup and release between two terminals. We must make a distinction
between direct call signaling and call signaling transferred via the gatekeeper.
Direct call signaling is used between two H.323 terminals either if no gatekeeper is available
in the network or if the gatekeeper requests this method in RAS signaling. Otherwise all
H.225 messages between two terminals are routed via the gatekeeper.
For historical reasons the call signaling following H.323 uses Q.931 messages as D-channel
ISDN signaling. At the start of H.323 an attempt was made to adapt standards and protocols
coming from ISDN and H.320.

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5.1.3 H.245 Call Control Signaling

H.245 contains control mechanisms for communication between H.323 terminals. It is

responsible for setup and tear down of logical communication channels. At the start
communication information about the technical capabilities of the terminals is exchanged. A
master/slave selection is performed. The sender encodes the signal into a real time data
stream. For decode the signal the receiver has to use the same codec as the sender. To
support bidirectional data transfer each terminal works as sender and receiver. Furthermore,
each terminal usually offers a variety of different codecs.
Thus signaling has make sure that both terminals use the same codec for data transmission.
This is achieved by capability exchange messages. The master/slave selection determines
which terminal - the master - controls the communication.
One example is the completion of a capability exchange where the master decides on an
audio or video codec to be used for data transmission. During a connection H.245 messages
detect if the opposite terminal is still working. The messages used are called keep-alive
messages.
If a connection is broken the working terminal informs the gatekeeper about the situation to
free the allocated resources.

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5.1.4 Audio and Video Codecs (1/2)

The support of voice transmission is obligatory for each H.323 system. H.323 contains
different standards for voice coding and compression. New coding procedures can be added
to the standard. The most important ones in H.323 are G.711, G.729A und G.723.1. All
H.323 terminals have to support G.711 coding, for historical reasons because G.711 is also
used for audio transmission in ISDN. For real time transmission of voice in WAN networks
G.723.1 or G.729A are often used.

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5.1.4 Audio and Video Codecs (2/2)

Videoconferenceing is an option in H.323 systems. If video transmission is required H.261

has to be implemented. H.261 is used in connections with data rates that are multiples of 64
kbps. Optional H.263 for video transmission via narrow band connections is possible. The
data rate used is agreed on during connection setup.

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5.1.5 T.120

The support of data conferencing in H.323 is also an option. It is realized by T.120 which is
part of H.323. T.120 contains standards supporting application sharing, whiteboard
functionalities (T.128) and file transfer (T.127) as used e.g. in MS Netmeeting
T.120 manages connection setup, reliable data transmission and terminal communication. It
supports multi-point conferences via different networks and connections like TCP/IP, IPX,
ISDN, and analog telephone networks.
T.120 may use H.225 to forward or receive data to and from the network.
T.120 may also format data itself and forward or receive it from the network.

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5.1.6 The RTP Protocol

The Real Time Protocol (RTP) is used for the end-to-end transport of real time data, like
audio and video data, via logical transport channels in IP networks.
In addition RTP offers classification, sequence numbering, time stamp, and delivery control
of the packages to be transmitted.
Synchronization and detection of package loss are possible using the timing information.
With the help of sequence numbers incoming packages are put in the correct order.
RTP is enhanced by Real Time Control Protocol (RTCP), this detects the delay time of a
voice connection for QoS monitoring and the dynamic adaptation of the Jitter-Buffers.

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5.1.7 H.235 Security

H.235 ensures security against eavesdropping especially in public IP networks. It includes

authentication, integrity check and ciphering.
In authentication the system checks that an end-point is that user or terminal he clains to be.
Both passwords and digital certificates may be used.
The integrity check helps detect manipulation of the transmitted data by using a check sum.
Ciphering guarantees the confidentiality of calls within an IP network.
As it may cause high delay times, ciphering is usual not used. A hardware ciphering solution
is often faster.
VPNs (Virtual Private Networks) are used for security when interconnecting private IP
networks via public networks. They are build between firewalls to transport voice and data in
a save tunnel.

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5.1.8 H.450 Zusatzdienste

An optional part of H.323 is H.450. It defines how supplementary services are implemented
in IP telephony. H.450 is constantly being enhanced and currently consist of 9 parts.

H.450.1 is generic protocol for support of supplementary services in H.323

H.450.2 handles call transfer
H.450.3 call forwarding
H.450.4 call waiting / call hold
H.450.5 call parking
H.450.6 call waiting
H.450.7 displays "message waiting"
H.450.8 caller identification
H.450.9 callback

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5.1.9 H.450 Supplementary Services

The translation function of an H.323 gateway - from packet switched IP network into a circuit
switched telephony network like ISDN - is described in H.246.
It is not only important to convert voice and video using the correct codec but also to adapt
the signaling e.g. call signaling might be adapted from H.225 into H.221, or call control
signaling from H.245 into H.230 for video conference in circuit switched ISDN.

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5.2 H.323 over IP based Networks (1/4)

The Internet Protocol IP offers the basic OSI layer 3 mechanisms for data packet transfer
within the Internet and non-public intranets. Being a simple, connectionless protocol it gives
routing information for networks consisting of different sub networks.
Each data package carries complete address information for reaching the target system. IP
guarantees neither complete delivery nor delivery in the correct sequence or without errors.
The scenario is similar to the serial input of letters into a letterbox.
It is certainly not guaranteed, in fact it is rather unlikely, that the letters will arrive at the
destination in the same order they were posted. Each letter is sent individually; during
transport there is no correlation between them. Thus it is a so-called connectionless
transmission. The demands on the underlying network are so low that any layer 2 protocol
may be used, ATM, frame relay, ISDN and Ethernet.

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5.2 H.323 over IP based Networks (2/4)

TCP is a reliable, connection oriented delivery service located on OSI layer 4 just on top of
IP. Each session is established by a 3-step handshake before any data is exchanged
between two hosts. Prior transmission data is segmented. The allocating a sequence number
to each segment provides reliability.
Thus giving the receiver the possibility to check if all parts have been received in when a
TCP segment is split in smaller parts. An acknowledgement "ACK" must send back form the
receiver to the transmitter within a certain time. Several segments may be acknowledged in
one go. If an acknowledgement is missing the data is retransmitted.
If a faulty segment is received the host ignores it and acts as low as missing, thus no "ACK"
is sent. Call signaling in VoIP uses TCP to ensure a reliable call setup.

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5.2 H.323 over IP based Networks (3/4)

UDP (User Datagram Protocol), a connectionless datagram service on OSI layer 4, offers a
non-reliable transmission. This means the the receipt of data or the correct sequence of
information is not ensured at the receiver. The advantage of UDP is the small header which
adds only a small amout of overhead.
Voice over IP uses UDP for voice transmission. Flow control and retransmission of real time
data as offered by TCP is not necessary.
As UDP is used for audio data, transmission continues with a data loss is 5% or 50%. If TCP
were used instead the delay caused by retransmission and acknowledgement would be
unacceptably high. In VoIP and other real time services a delay time less of than 150 ms is
more important than reliable package transport.

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5.2 H.323 over IP based Networks (4/4)

In the OSI model Internet Protocol corresponds to the network layer and Transmission
Control Protocol to the transport layer. As we can see, the latter layer is split into a reliable
and a non-reliable transport service using TCP and UDP respectively. It is important to
remember that neither of these two layers nor the underlying network is covered by H.323.
H.323 uses TCP and UDP for data transmission in IP based networks.
Q.931 call signaling, H.245 control information and T.120 application data have to be
transmitted completely and without errors so H.323 uses connection oriented TCP. Real time
data has to arrive at the receiver within a certain time. A data loss of about 5% is acceptable.
Audio, video and real time application data as well as RAS signaling use UDP. For multipoint
conferences H.323 uses UDP, IP multicast and RTP.
In connectionless transmission via UDP data reassembling is shifted to a higher layer
protocol. So a second protocol, the Real Time Protocol (RTP), enables resynchronization of
data at the receiver using sequence numbers and a time stamp in combination with a jitter
buffer.

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