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NEXT GENERATION NETWORK
Softswitch
A softswitch is a central device in a telephone network which connects calls
from one phone line to another, entirely by means of software running on a
computer system. This work was formerly carried out by hardware, with
physical switchboards to route the calls.
A softswitch is typically used to control connections at the junction point
between circuit and packet networks. A single device containing both the
switching logic and the switching fabric can be used for this purpose;
however, modern technology has led to a preference for decomposing this
device into a Call Agent and a Media Gateway.
The Call Agent takes care of functions like billing, call routing, signalling, call
services and so on and is the 'brains' of the outfit. A Call Agent may control
several different Media Gateways in geographically dispersed areas over a
TCP/IP link.
The Media Gateway connects different types of digital media stream together
to create an end-to-end path for the media (voice and data) in the call. It may
have interfaces to connect to traditional PSTN networks like DS1 or DS3 ports
(E1 or STM1 in the case of non-US networks), it may have interfaces to
connect to ATM and IP networks and in the modern system will have Ethernet
interfaces to connect VoIP calls. The call agent will instruct the media
gateway to connect media streams between these interfaces to connect the
call - all transparently to the end-users.
The softswitch generally resides in a building owned by the telephone
company called a central office. The central office will have telephone trunks
to carry calls to other offices owned by the telephone company and to other
telephone companies (aka the Public Switched Telephone Network or PSTN).
Looking towards the end users from the switch, the Media Gateway may be
connected to several access devices. These access devices can range from
small Analog Telephone Adaptors (ATA) which provide just one RJ11
telephone jack to an Integrated Access Device (IAD) or PBX which may
provide several hundred telephone connections.
Typically the larger access devices will be located in a building owned by the
telephone company near to the customers they serve. Each end user can be
connected to the IAD by a simple pair of copper wires.
The medium sized devices and PBXs will typically be used in a business
premises and the single line devices would probably be found in residential
premises.
In more recent times (i.e., the IP Multimedia Subsystem or IMS), the
Softswitch element is represented by the Media Gateway Controller (MGC)
element, and the term "Softswitch" is rarely used in the IMS context.
Architecture
The distributed system is composed of a Call Agent (or Media Gateway
Controller), at least one Media Gateway (MG) that performs the conversion of
media signals between circuits and packets, and at least one Signaling
gateway (SG) when connected to the PSTN.
The Call Agent uses MGCP to tell the Media Gateway:
Typically, a Media Gateway is configured with a list of Call Agents from which
it may accept programming (where that list normally comprises only one or
two Call Agents). In principle, event notifications may be sent to different Call
Agents for each endpoint on the gateway (as programmed by the Call Agents,
by setting the Notified Entity parameter). In practice however, it is usually
desirable that at any given moment all endpoints on a gateway should be
controlled by the same Call Agent; other Call Agents are available only to
provide redundancy in the event that the primary Call Agent fails, or loses
contact with the Media Gateway. In the event of such a failure it is the backup
Call Agent's responsibility to reprogram the MG so that the gateway comes
under the control of the backup Call Agent. Care is needed in such cases; two
Call Agents may know that they have lost contact with one another, but this
does not guarantee that they are not both attempting to control the same
gateway. The ability to audit the gateway to determine which Call Agent is
currently controlling can be used to resolve such conflicts.
MGCP assumes that the multiple Call Agents will maintain knowledge of
device state among themselves (presumably with an unspecified protocol) or
rebuild it if necessary (in the face of catastrophic failure). Its failover features
take into account both planned and unplanned outages.
Protocol Overview
MGCP packets are unlike what you find in many other protocols. Usually
wrapped in UDP port 2427, the MGCP datagrams are formatted with
whitespace, much like you would expect to find in TCP protocols. An MGCP
packet is either a command or a response.
Commands begin with a four-letter verb. Responses begin with a three
number response code.
There are eight (8) command verbs:
Two verbs are used by a Call Agent to query (the state of) a Media Gateway:
One verb is used by a Media Gateway to indicate to the Call Agent that it has
detected an event for which the Call Agent had previously requested
notification of (via the RQNT command verb):
NTFY - Notify
One verb is used by a Media Gateway to indicate to the Call Agent that it is in
the process of restarting:
H.323
H.323 is an umbrella Recommendation from the ITU Telecommunication
Standardization Sector (ITU-T) that defines the protocols to provide audio-
visual communication sessions on any packet network.
It is widely implemented by voice and videoconferencing equipment
manufacturers, is used within various Internet real-time applications such as
GnuGK, NetMeeting and X-Meeting, and is widely deployed worldwide by
service providers and enterprises for both voice and video services over
Internet Protocol (IP) networks.
It is a part of the ITU-T H.32x series of protocols, which also address
multimedia communications over Integrated Services Digital Network (ISDN),
Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7),
and 3G mobile networks.
H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol
and is suited for transmitting calls across networks using a mixture of IP,
PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call
model, eases the introduction of IP telephony into existing networks of ISDN-
based PBX systems, including transitions to IP-based Private Branch
eXchanges (PBXs).
Within the context of H.323, an IP-based PBX might be an H.323 Gatekeeper
or other call control element that provides service to telephones or
videophones. Such a device may provide or facilitate both basic services and
supplementary services, such as call transfer, park, pick-up, and hold.
While H.323 excels at providing basic telephony functionality and
interoperability, H.323’s strength lies in multimedia communication
functionality designed specifically for IP networks.
Contents
1 History
2 Protocols
3 Codecs
4 H.323 Architecture
4.1.1 Terminals
4.1.3 Gateways
4.1.4 Gatekeepers
5 Use cases
History
The first version of H.323 was published by the ITU in November 1996 with
an emphasis of enabling videoconferencing capabilities over a Local Area
Network (LAN), but was quickly adopted by the industry as a means of
transmitting voice communication over a variety of IP networks, including
WANs and the Internet (see VoIP).
Over the years, H.323 has been revised and re-published with enhancements
necessary to better-enable both voice and video functionality over Packet-
switched networks, with each version being backward-compatible with the
previous version. Recognizing that H.323 was being used for communication,
not only on LANs, but over WANs and within large carrier networks, the title of
H.323 was changed when published in 1998. The title, which has since
remained unchanged, is "Packet-Based Multimedia Communications
Systems." The current version of H.323, commonly referred to as "H.323v6",
was published in 2006.
One strength of H.323 was the relatively early availability of a set of
standards, not only defining the basic call model, but also the supplementary
services needed to address business communication expectations.
H.323 was the first VoIP standard to adopt the Internet Engineering Task
Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio
and video over IP networks.
Protocols
H.323 is a system specification that describes the use of several ITU-T and
IETF protocols. The protocols that comprise the core of almost any H.323
system are:
H.235 series describes security within H.323, including security for both
signaling and media.
H.239 describes dual stream use in videoconferencing, usually one for live
video, the other for still images.
Codecs
H.323 utilizes both ITU-defined codecs and codecs defined outside the ITU.
Codecs that are widely implemented by H.323 equipment include:
H.323 Architecture
The H.323 system defines several network elements that work together in
order to deliver rich multimedia communication capabilities. Those elements
are Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers, and
Border Elements. Collectively, terminals, multipoint control units and
gateways are often referred to as endpoints.
While not all elements are required, at least two terminals are required in
order to enable communication between two people. In most H.323
deployments, a gatekeeper is employed in order to, among other things,
facilitate address resolution.
Terminals
most H.323 systems do not implement such a wide array of capabilities, but
the logical arrangement is useful in understanding the relationships.
Gateways
Gateways are devices that enable communication between H.323 networks
and other networks, such as PSTN or ISDN networks. If one party in a
conversation is utilizing a terminal that is not an H.323 terminal, then the call
must pass through a gateway in order to enable both parties to communicate.
Gateways are widely used today in order to enable the legacy PSTN phones
to interconnect with the large, international H.323 networks that are presently
deployed by services providers. Gateways are also used within the enterprise
in order to enable enterprise IP phones to communicate through the service
provider to users on the PSTN.
Gateways are also used in order to enable videoconferencing devices based
on H.320 and H.324 to communicate with H.323 systems. Most of the third
generation (3G) mobile networks deployed today utilize the H.324 protocol
and are able to communicate with H.323-based terminals in corporate
networks through such gateway devices.
Gatekeepers
A Gatekeeper is an optional component in the H.323 network that provides a
number of services to terminals, gateways, and MCU devices. Those services
include endpoint registration, address resolution, admission control, user
authentication, and so forth. Of the various functions performed by the
gatekeeper, address resolution is the most important as it enables two
endpoints to contact each other without either endpoint having to know the IP
address of the other endpoint on.
Gatekeepers may be designed to operate in one of two signaling modes,
namely "direct routed" and "gatekeeper routed" mode. Direct routed mode is
the most efficient and most widely deployed mode. In this mode, endpoints
utilize the RAS protocol in order to learn the IP address of the remote
endpoint and a call is established directly with the remote device. In the
gatekeeper routed mode, call signaling always passes through the
gatekeeper. While the latter requires the gatekeeper to have more processing
power, it also gives the gatekeeper complete control over the call and the
ability to provide supplementary services on behalf of the endpoints.
Alerting
Information
Release Complete
Facility
Progress
Status and Status Inquiry Notify
Figure 3 - Establishment of an H.323 call
In the simplest form, an H.323 call may be established as follows (figure 3):
In this example, the endpoint (EP) on the left initiated communication with the
gateway on the right and the gateway connect the call with the called party. In
reality, call flows are often more complex than the one shown, but most calls
that utilize the Fast Connect procedures defined within H.323 can be
established with as few as 2 or 3 messages. Endpoints must notify their
gatekeeper (if gatekeepers are used) that they are in a call.
Once a call has concluded, a device will send a Release Complete message.
Endpoints are then required to notify their gatekeeper (if gatekeepers are
used) that the call has ended.
RAS Signaling
Endpoints use the RAS protocol in order to communicate with a gatekeeper.
Likewise, gatekeepers use RAS to communicate with peer gatekeepers. RAS
is a fairly simple protocol comprised of just a few messages. Namely:
Fast Connect
Figure 5 - A typical H.245 exchange
Use cases
H.323 and Voice over IP services
Voice over Internet Protocol (VoIP) describes the transmission of voice using
the Internet or other packet switched networks. ITU-T Recommendation
H.323 is one of the standards used in VoIP. VoIP requires a connection to the
Internet or another packet switched network, a subscription to a VoIP service
provider and a client (an analogue telephone adapter (ATA), VoIP Phone or
"soft phone"). The service provider offers the connection to other VoIP
services or to the PSTN. Most service providers charge a monthly fee, then
additional costs when calls are made.[1] Using VoIP between two enterprise
locations would not necessarily require a VoIP service provider, for example.
H.323 has been widely deployed by companies who wish to interconnect
remote locations over IP using a number of various wired and wireless
technologies.
H.323 and Videoconference services
A videoconference, or video teleconference (VTC), is a set of
telecommunication technologies allowing two or more locations to interact via
two-way video and audio transmissions simultaneously. There are basically
two types of videoconferencing; dedicated VTC systems have all required
components packaged into a single piece of equipment while desktop VTC
systems are add-ons to normal PC's, transforming them into VTC devices.
Simultaneous videoconferencing among three or more remote points is
possible by means of a Multipoint Control Unit (MCU). There are MCU
bridges for IP and ISDN-based videoconferencing. Due to the price point and
proliferation of the Internet, and broadband in particular, there has been a
strong spurt of growth and use of H.323-based IP videoconferencing. H.323 is
accessible to anyone with a high speed Internet connection, such as DSL.
Videoconferencing is utilized in various situations, for example; distance
education, telemedicine and business.[2]
International Conferences
H.323 has been used in the industry to enable large-scale international video
conferences that are significantly larger than the typical video conference.
One of the most widely attended is an annual event called “Megaconference”.
The Mega conferences are special non-profit world-wide events which use the
H.323 protocol to create a virtual conference involving hundreds of locations
and thousands of people. Everyone in the world with H.323 equipment is
invited to participate. They are the world’s largest video conferences. The first
Mega conference was held in 1999, and it has been held annually ever since.
The Mega conferences are run as professional conferences, with no central
location. There are presentations (called Interactions) by users of H.323
technology, vendor presentations, roll calls, musical events and open periods
called mega conference Cafes where anyone can talk to anyone.A particularly
popular portion is the Roll Calls, where all registrants are given a moment to
say hello to the world; they can say whetever they wish, sing a song, play a
video or whatever. A network of 30 or so MCUs is created for the event, all
cascaded together. Background chats are run for the presenters, the MCU
managers and the audience, to coordinate the event in real-time. The event is
also streamed out to the world, and is recorded for later distribution on
DVDs.[3] There have been a number of spinoffs of the Mega conference,
beginning with Mega conference Jr, which started in 2002. That event is
intended for students of all ages, and students make all the presentations.[4]
The Mega conferences and their spin-offs received the first-ever Internet2
Driving Exemplary Applications award in 2006
SIGTRAN
SIGTRAN is the name given to an Internet Engineering Task Force (IETF)
working group that produced specifications for a family of protocols that
provide reliable datagram service and user layer adaptations for SS7 and
ISDN communications protocols. SIGTRAN is logically an extension of the
SS7 protocol family. It supports the same application and call management
paradigms as SS7 but uses an IP transport called Stream Control
Transmission Protocol (SCTP) as its underlying transport vehicle. Indeed, the
most significant protocol defined by the SIGTRAN group was SCTP, which is
used to carry PSTN signaling over IP.
The SIGTRAN group was significantly influenced by telecommunications
engineers intent on using the new protocols for adapting VoIP networks to the
PSTN with special regard to signaling applications. Recently, SCTP is finding
applications beyond its original purpose wherever reliable datagram service is
desired.
The SIGTRAN family of protocols includes:
Network refers to the fact that ISDN is not simply a point-to-point solution like a
leased line. ISDN networks extend from the local telephone exchange to the
remote user and includes all of the telecommunications and switching equipment
in between.
The purpose of the ISDN is to provide fully integrated digital services to the
users. These services fall under three categories: bearer services,
supplementary services and teleservices.
5. Application layer
4. Transport layer
TCP · UDP · DCCP · SCTP · RSVP · ECN ·
(more)
3. Network/internet layer
IP (IPv4 · IPv6) · OSPF · IS-IS · BGP ·
IPsec · ARP · RARP · RIP · ICMP · ICMPv6
· IGMP · (more)
2. Data link layer
802.11 (WLAN) · 802.16 · Wi-Fi · WiMAX ·
ATM · DTM · Token ring · Ethernet · FDDI ·
Frame Relay · GPRS · EVDO · HSPA ·
HDLC · PPP · PPTP · L2TP · ISDN ·
ARCnet · LLTD · (more)
1. Physical layer
Ethernet physical layer · Modems · PLC ·
SONET/SDH · G.709 · Optical fiber ·
Coaxial cable · Twisted pair · (mo
MTP is made up of three levels, corresponding to layers in the OSI model: MTP
Level 1 corresponds to OSI Layer 1 (the physical layer), MTP Level 2 to OSI
Layer 2 (the data link layer), and MTP Level 3 to OSI Layer 3 (the network layer).
MTP Level 3 is usually abbreviated as MTP3. Likewise MTP Level 2 and MTP
Level 1 are abbreviated as MTP2 and MTP1.
MTP1 represents the physical layer. That is, the layer that is responsible for the
connection of SS7 Signaling Points into the transmission network over which
they communicate with each other. Primarily, this involves the conversion of
messaging into electrical signal and the maintenance of the physical links
through which these pass. In this way, it is analogous to the Layer 1 of ISDN or
other, perhaps more familiar, protocols.
for MTP3. These tests are used to validate the correct implementation of the
MTP protocol.
Different countries use different variants of the MTP protocols. In North America,
the formal standard followed is the Telcordia Technologies (formerly Bellcore)
document GR-246-CORE
Layer Protocols
Transport SCCP
Network MTP Level 3
Data link MTP Level 2 ...
Physical MTP Level 1
The Message Transfer Part (MTP) is part of the Signaling System 7 (SS7) used
for communication in Public Switched Telephone Networks. MTP is responsible
for reliable, unduplicated and in-sequence transport of SS7 messages between
communication partners.
MTP is made up of three levels, corresponding to layers in the OSI model: MTP
Level 1 corresponds to OSI Layer 1 (the physical layer), MTP Level 2 to OSI
Layer 2 (the data link layer), and MTP Level 3 to OSI Layer 3 (the network layer).
MTP Level 3 is usually abbreviated as MTP3. Likewise MTP Level 2 and MTP
Level 1 are abbreviated as MTP2 and MTP1.
MTP1 represents the physical layer. That is, the layer that is responsible for the
connection of SS7 Signaling Points into the transmission network over which
they communicate with each other. Primarily, this involves the conversion of
messaging into electrical signal and the maintenance of the physical links
through which these pass. In this way, it is analogous to the Layer 1 of ISDN or
other, perhaps more familiar, protocols.
Different countries use different variants of the MTP protocols. In North America,
the formal standard followed is the Telcordia Technologies (formerly Bellcore)
document GR-246-CORE.
M3UA
The introduction to this article provides insufficient context for those unfamiliar with the subject.
Please help improve the article with a good introductory style.
Layer Protocols
Transport SCCP
Network MTP Level 3
Data link MTP Level 2 ...
MTP-3
M3UA stands for MTP Level 3 (MTP3) User Adaptation Layer as defined by the
IETF SIGTRAN working group. M3UA enables the SS7 protocol's User Parts
(e.g. ISUP, SCCP and TUP) to run over IP instead of telephony equipment like
ISDN and PSTN. It is recommended to use the services of SCTP to transmit
M3UA.
5. Application layer
4. Transport layer
TCP · UDP · DCCP · SCTP · RSVP · ECN ·
(more)
3. Network/internet layer
IP (IPv4 · IPv6) · OSPF · IS-IS · BGP ·
IPsec · ARP · RARP · RIP · ICMP · ICMPv6
· IGMP · (more)
2. Data link layer
802.11 (WLAN) · 802.16 · Wi-Fi · WiMAX ·
ATM · DTM · Token ring · Ethernet · FDDI ·
Frame Relay · GPRS · EVDO · HSPA ·
HDLC · PPP · PPTP · L2TP · ISDN ·
ARCnet · LLTD · (more)
1. Physical layer
Ethernet physical layer · Modems · PLC ·
SONET/SDH · G.709 · Optical fiber ·
Coaxial cable · Twisted pair · (more)
Contents
1 Message-based multi-streaming
2 Benefits
3 Motivations
4 Comparison between transport
layers
5 Implementations
6 Packet structure
7 See also
8 External links
Message-based multi-streaming
TCP ensures the correct order of bytes in the stream by conceptually assigning a
sequence number to each byte sent and ordering these bytes based on that
sequence number when they arrive. SCTP, on the other hand, assigns different
sequence numbers to messages sent in a stream. This allows independent
ordering of messages in different streams. However, message ordering is
optional in SCTP. If the user application so desires, messages will be processed
in the order they are received instead of the order they were sent, should these
differ.
Benefits
the SS7 signaling network in IP. This IETF effort is known as SIGTRAN. In the
meantime, other uses have been proposed, for example the Diameter protocol
and Reliable server pooling ("RSerPool").
Motivations
Transmission Control Protocol (TCP) has provided the primary means to transfer
data across the Internet in a reliable way. However, TCP has imposed limitations
on several applications. From
TCP provides both reliable data transfer and strict order-of- transmission delivery
of data. Some applications need reliable transfer without sequence maintenance,
while others would be satisfied with partial ordering of the data. In both of these
cases, the head-of-line blocking offered by TCP causes unnecessary delay.
The limited scope of TCP sockets complicates the task of providing highly-
available data transfer capability using multi-homed hosts.
Preserve message
Yes No Yes
boundary
Bundling No No Yes
Layer Protocols
Transport SCCP
Network MTP Level 3
Data link MTP Level 2 ...
Physical MTP Level 1 ...
Contents
1 Published specification
2 Routing facilities beyond MTP-3
3 Classes of service
3.1 Class 0: Basic connectionless
3.2 Class 1: Sequenced
connectionless
3.3 Class 2: Basic connection-
oriented
3.4 Class 3: Flow control
connection oriented
4 Transport over IP Networks
Published specification
Although MTP-3 provides routing capabilities based upon the Point Code, SCCP
allows routing using a Point Code and Subsystem number or a Global Title.
Address Indicator
Subsystem indicator: The address includes a Subsystem Number
Point Code indicator: The address includes a Point Code
Global title indicator
No Global Title
Global Title includes Translation Type (TT), Numbering Plan
Indiciator (NPI) and Type of Number (TON)
Global Title includes Translation Type only
Routing indicator
Route using Global Title only
Route using Point Code/Subsystem number
Address Indicator Coding
Address Indicator coded as national (the Address Indicator is
treated as international if not specified)
Classes of service
to transfer in the "data" field of XUDT (or as a network option LUDT) messages.
At the destination node, the NSDU is reassembled.
In the SIGTRAN suite of protocols, there are two primary methods of transporting
SCCP applications across Internet Protocol networks: SCCP can be transported
directly using the MTP level 3 User Adaptation protocol (M3UA), a protocol which
provides support for users of MTP-3—including SCCP. Alternatively, SCCP
applications can operate over the SCCP User Adaptation protocol (SUA) which is
a form of modified SCCP designed specifically for use in IP networking.
V5 interface
The introduction to this article provides insufficient context for those unfamiliar
with the subject.
Please help improve the article with a good introductory style.
Softswitch
A softswitch is a central device in a telephone network which connects calls
from one phone line to another, entirely by means of software running on a
computer system. This work was formerly carried out by hardware, with physical
switchboards to route the calls.
The Call Agent takes care of functions like billing, call routing, signalling, call
services and so on and is the 'brains' of the outfit. A Call Agent may control
several different Media Gateways in geographically dispersed areas over a
TCP/IP link.
The Media Gateway connects different types of digital media stream together to
create an end-to-end path for the media (voice and data) in the call. It may have
interfaces to connect to traditional PSTN networks like DS1 or DS3 ports (E1 or
STM1 in the case of non-US networks), it may have interfaces to connect to ATM
and IP networks and in the modern system will have Ethernet interfaces to
connect VoIP calls. The call agent will instruct the media gateway to connect
media streams between these interfaces to connect the call - all transparently to
the end-users.
Looking towards the end users from the switch, the Media Gateway may be
connected to several access devices. These access devices can range from
small Analog Telephone Adaptors (ATA) which provide just one RJ11 telephone
jack to an Integrated Access Device (IAD) or PBX which may provide several
hundred telephone connections.
Typically the larger access devices will be located in a building owned by the
telephone company near to the customers they serve. Each end user can be
connected to the IAD by a simple pair of copper wires.
The medium sized devices and PBXs will typically be used in a business
premises and the single line devices would probably be found in residential
premises.
In more recent times (i.e., the IP Multimedia Subsystem or IMS), the Softswitch
element is represented by the Media Gateway Controller (MGC) element, and
the term "Softswitch" is rarely used in the IMS context.
The feature server, often built into a call agent/softswitch, is the functional
component that provides call-related features. Capabilities such as call
forwarding, call waiting, and last call return, if implemented in the network, are
implemented in the feature server. The feature server works closely with the call
agent, and may call upon the media server to provide these services. These
features do not require the subscriber to explicitly request them but tend to be
triggered within the call handling logic.
An example of a feature service is last call return, in which the user picks up the
phone, dials *69, and hears, “The number that last called you was xxx-xxx-xxxx.
Press 1 to return this call.” When the call agent sees the dial string *69, it triggers
an invocation of the feature server function. The feature server examines its
database, finds the user and the caller identification of the last call, then asks the
media server to play the announcement and collect a digit. When the media
server returns a “1”, the feature server instructs the call agent to establish a call
between the user and the party that last called that user.
RTP PROTOCOL
The Real-time Transport Protocol (or RTP) defines a standardized packet
format for delivering audio and video over the Internet. It was developed by
the Audio-Video Transport Working Group of the IETF and first published in
1996 which was made obsolete in 2003.Real time transport protocol can also
be used in conjunction with RTSP protocol which enhances the field of
multimedia applications.
RTP does not have a standard TCP or UDP port on which it communicates.
The only standard that it obeys is that UDP communications are done via an
even port and the next higher odd port is used for RTP Control Protocol
(RTCP) communications. Although there are no standards assigned, RTP is
generally configured to use ports 16384-32767. RTP can carry any data with
real-time characteristics, such as interactive audio and video. Call setup and
tear-down for VoIP applications is usually performed by either SIP or H.323
protocols. The fact that RTP uses a dynamic port range makes it difficult for it
to traverse firewalls. In order to get around this problem, it is often necessary
to set up a STUN server.
It was originally designed as a multicast protocol, but has since been applied
in many unicast applications. It is frequently used in streaming media systems
(in conjunction with RTSP) as well as videoconferencing and push to talk
systems (in conjunction with H.323 or SIP), making it the technical foundation
of the Voice over IP industry. It goes along with the RTCP and is built on top
of the User Datagram Protocol (UDP). Applications using RTP are less
sensitive to packet loss, but typically very sensitive to delays, so UDP is a
better choice than TCP for such applications.
The services provided by RTP include:
Contents
[hide]
1 Packet structure
3 Mathematical background
9-
+ Bits 0-1 2 3 4-7 8 16-31
15
Sequence
0 Ver. P X CC M PT
Number
32 Timestamp
64 SSRC identifier
96 ... CSRC identifiers ...
96+(CC×32) Extension header (optional).
96+(CC×32)
+ Data
(X×((EHL+1)×32))
Packet structure
The RTP header size is 12 bytes.
Ver.
(2 bits) Indicates the version of the protocol. Current version is 2.
P
(1 bit) Used to indicate if there are extra padding bytes at the end of the
RTP packet.
X
(1 bit) Indicates if the extensions to the protocol are being used in the
packet.
CC
(4 bits) Contains the number of CSRC identifiers that follow the fixed
header.
M
(1 bit) Used at the application level and is defined by a profile. If it is
set, it means that the current data has some special relevance for the
application.
PT
(7 bits) Indicates the format of the payload and determines its
interpretation by the application.
SSRC
Indicates the synchronization source.
CSRC
Contributing source ID.
Extension header
Indicates the length of the extension (EHL=extension header length) in
32bit units, excluding the 32bits of the extension header.
There are several type of RTCP packets: Sender report packet, Receiver
report packet, Source Description RTCP Packet, Goodbye RTCP Packet and
Application Specific RTCP packets.
RTCP itself does not provide any flow encryption or authentication means.
SRTCP protocol can be used for that purpose.
The Session Initiation Protocol (SIP) is a signalling protocol, widely used for
setting up and tearing down multimedia communication sessions such as
voice and video calls over the Internet. Other feasible application examples
include video conferencing, streaming multimedia distribution, instant
messaging, presence information and online games. In November 2000, SIP
was accepted as a 3GPP signaling protocol and permanent element of the
IMS architecture for IP based streaming multimedia services in cellular
systems.
The protocol can be used for creating, modifying and terminating two-party
(unicast) or multiparty (multicast) sessions consisting of one or several media
streams. The modification can involve changing addresses or ports, inviting
more participants, adding or deleting media streams, etc.
The SIP protocol is situated at the session layer in the OSI model, and at the
application layer in the TCP/IP model. SIP is designed to be independent of
the underlying transport layer; it can run on TCP, UDP, or SCTP. It was
originally designed by Henning Schulzrinne (Columbia University) and Mark
Handley (UCL) starting in 1996.
SIP has the following characteristics:
1 Protocol design
4 Conformance testing
5 Commercial applications
Protocol design
SIP clients typically use TCP or UDP (typically on port 5060) to connect to
SIP servers and other SIP endpoints. SIP is primarily used in setting up and
tearing down voice or video calls. However, it can be used in any application
where session initiation is a requirement. These include Event Subscription
and Notification, Terminal mobility and so on. There are a large number of. All
voice/video communications are done over separate session protocols,
typically RTP.
A motivating goal for SIP was to provide a signaling and call setup protocol for
IP-based communications that can support a superset of the call processing
functions and features present in the public switched telephone network
(PSTN). SIP by itself does not define these features; rather, its focus is call-
setup and signaling. However, it has been designed to enable the building of
such features in network elements known as Proxy Servers and User Agents.
These are features that permit familiar telephone-like operations: dialing a
number, causing a phone to ring, hearing ringback tones or a busy signal.
Implementation and terminology are different in the SIP world but to the end-
user, the behavior is similar.
SIP-enabled telephony networks can also implement many of the more
advanced call processing features present in Signaling System 7 (SS7),
though the two protocols themselves are very different. SS7 is a centralized
protocol, characterized by a complex central network architecture and dumb
endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol. As
such it requires only a simple (and thus scalable) core network with
intelligence distributed to the network edge, embedded in endpoints
(terminating devices built in either hardware or software). SIP features are
implemented in the communicating endpoints (i.e. at the edge of the network)
as opposed to traditional SS7 features, which are implemented in the
network.
Although many other VoIP signaling protocols exist, SIP is characterized by
its proponents as having roots in the IP community rather than the telecom
industry. SIP has been standardized and governed primarily by the IETF while
the H.323 VoIP protocol has been traditionally more associated with the ITU.
However, the two organizations have endorsed both protocols in some
fashion.
SIP works in concert with several other protocols and is only involved in the
signaling portion of a communication session. SIP acts as a carrier for the
Session Description Protocol (SDP), which describes the media content of the
session, e.g. what IP ports to use, the codec being used etc. In typical use,
SIP "sessions" are simply packet streams of the Real-time Transport Protocol
(RTP). RTP is the carrier for the actual voice or video content itself.
SIP is similar to HTTP and shares some of its design principles: It is human
readable and request-response structured. SIP shares many HTTP status
codes, such as the familiar '404 not found'. SIP proponents also claim it to be
simpler than H.323. However, some would counter that while SIP originally
SIP also requires proxy and registrar network elements to work as a practical
service. Although two SIP endpoints can communicate without any
intervening SIP infrastructure, which is why the protocol is described as peer-
to-peer, this approach is impractical for a public service. There are various
implementations that can act as proxy and registrar.
"SIP makes use of elements called proxy servers to help route
requests to the user's current location, authenticate and authorize
users for services, implement provider call-routing policies, and provide
features to users."
"SIP also provides a registration function that allows users to upload
their current locations for use by proxy servers. "
"Since registrations play an important role in SIP, a User Agent Server
that handles a REGISTER is given the special name registrar."
"It is an important concept that the distinction between types of SIP
servers is logical, not physical."
Commercial applications
Firewalls typically block media packet types such as UDP, though one way
around this is to use TCP tunnelling and relays for media in order to provide
NAT and firewall traversal. One solution involves tunnelling the media packets
within TCP or HTTP packets to a relay. This solution uses additional
functionality in conjunction with SIP, and packages the media packets into a
TCP stream which is then sent to the relay. The relay then extracts the
packets and sends them on to the other endpoint. If the other endpoint is
behind a symmetrical NAT, or corporate firewall that does not allow VoIP
traffic, the relay would transfer the packets to another tunnel. One
disadvantage of this approach is that TCP was not designed for real time
traffic such as voice, so an optimized form of the protocol is sometimes used.
As envisioned by its originators, SIP's peer-to-peer nature does not enable
network-provided services. For example, the network can not easily support
legal interception of calls (referred to in the United States by the law
governing wiretaps, CALEA). Emergency calls (calls to E911 in the USA) are
difficult to route. It is difficult to identify the proper Public Service Answering
Point, PSAP because of the inherent mobility of IP end points and the lack of
any network location capability.
Many VoIP phone companies allow customers to bring their own SIP devices,
as SIP-capable telephone sets, or softphones. The new market for consumer
SIP devices continues to expand.
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