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TextBook: Soft Switch

ETE405 :: Lecture 15 Chapter 3 Softswitch Architecture p.60- p.


Differe ce !etwee TC" a # $D" "rotoco%

TCP(Transmission Control Protocol). UDP(User Datagram Protocol). A simpler TCP is a connection-oriented protocol, a message-based connectionless protocol. connection can be made from client to ith UDP yo! send messages(pac"ets) server, and from then on any data can be across the net#or" in ch!n"s. sent along that connection.

$eliable % #hen yo! send a message along a TCP soc"et, yo! "no# it #ill get there !nless the connection fails completely. &f it gets lost along the #ay, the server #ill re-re'!est the lost part. This means complete integrity, things don(t get corr!pted.

Unreliable % hen yo! send a message, yo! don(t "no# if it(ll get there, it co!ld get lost on the #ay.


)rdered % if yo! send t#o messages +ot ordered % &f yo! send t#o messages along a connection, one after the other, o!t, yo! don(t "no# #hat order they(ll yo! "no# the first message #ill get there arrive in. first. *o! don(t have to #orry abo!t data arriving in the #rong order. ,eavy#eight % #hen the lo# level parts of the TCP -stream. arrive in the #rong order, resend re'!ests have to be sent, and all the o!t of se'!ence parts have to be p!t bac" together, so re'!ires a bit of #or" to piece together. /ight#eight % +o ordering of messages, no trac"ing connections, etc. &t(s 0!st fire and forget1 This means it(s a lot '!ic"er, and the net#or" card 2 )3 have to do very little #or" to translate the data bac" from the pac"ets.


Signaling Protocols

Signaling establishes the virtual circuit over the network for that media stream. Signaling is independent of the media flow. It determines the type of media to be used in a call. Signaling is concurrent throughout the call. Two types of signaling are currently popular in VoIP:

H.323 SIP

Signaling and transport protocols used in VoIP


H.323 is the International Telecommunication UnionTelecommunications Standardization Sector (ITU-T) recommendation for packetbased multimedia communication. H.323 was developed before the emergence of VoIP. As it was not specifically designed for VoIP, it has faced a good deal of competition from a competing protocol, SIP, which was designed specifically for VoIP.

The H.323 standard is a cornerstone technology for the transmission of real-time audio, video, and data communications over packet-based networks. It specifies the components, protocols, and procedures providing multimedia communication over packet-based networks. H.323 can be applied in a variety of mechanisms: audio only (IP telephony); audio and video (videotelephony); audio and data; and audio, video, and data.

Interworking with Other Multimedia Networks

The H.323 standard specifies four kinds of components, which, when networked together, provide the point-to-point and point-to-multipoint multimedia communication services:

Terminals Gateways Gatekeepers multipoint control units (MCUs).


Used for real-time bidirectional multimedia communications, an H.323 terminal can either be a personal computer (PC) or a stand-alone device running an H.323 and the multimedia applications.


A gatekeeper can be considered the brain of the H.323 network. It is the focal point for all calls within the H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization, and authentication of terminals and gateways, bandwidth management, accounting, billing, and charging. Gatekeepers may also provide call-routing services.

Multipoint Control Units (MCUs)

MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video coder/decoder (codec) to use, and may handle the media stream.

&essa'e (ea#ers

)ou use *essa'e hea#ers to specif+ the ca%%i ' part+, ca%%e# part+, route, a # *essa'e t+pe of a ca%%. The four 'roups of *essa'e hea#ers are as fo%%ows:

-e era% hea#ers.App%+ to re/uests a # respo ses. 0 tit+ hea#ers.Defi e i for*atio a!out the *essa'e !o#+ t+pe a # %e 'th. 1e/uest hea#ers.0 a!%e the c%ie t to i c%u#e a##itio a% re/uest i for*atio . 1espo se hea#ers.0 a!%e the ser2er to i c%u#e a##itio a% respo se i for*atio .

These *ai hea#er 'roups, a%o ' with 33 correspo #i ' hea#ers, are %iste# i the fo%%owi ' ta!%e.


0xp%a atio s for So*e 4e+ S5" (ea#ers

4 4 4 4 4

To: 5#e tifies the recipie t of the re/uest. 6ro*: 5 #icates the i itiator of the re/uest. Su!7ect: Descri!es the ature of the ca%%. 8ia: 5 #icates the path take !+ the re/uest. Ca%%-5D: $ i/ue%+ i#e tifies a specific i 2itatio or a%% re'istratio s of a specific c%ie t. Co te t-9e 'th: 5#e tifies the si:e of the *essa'e !o#+ i octets. Co te t-T+pe: 5 #icates the *e#ia t+pe of the *essa'e !o#+. 0xpires: 5#e tifies the #ate a # ti*e whe the *essa'e co te t expires. 1oute: 5 #icates the route take !+ a re/uest.

0xa*p%e of a *essa'e hea#er

8ia: S5"/;.0/$D" 1<;.16=.0.100:5060>rport>!ra ch?:<h-@!46@6@6@1000000033@3c5; 63<0000;0a600000e@5 Co te t-9e 'th: 0 Ca%%-5D: <11D3;05-00D6-@53;-B0B;-61B;<@6360==A1<;.16=.0.100 CSe/: 1 AC4 6ro*: B"rue!aBCsip:;0000A*iasterisk.co*D>ta'?=<;;@0@61@6=; &ax-6orwar#s: 30 1oute: Csip:;0001A1<;.16=.0.1D To: Csip:;0001A*iasterisk.co*D>ta'?as0a;3!<;= $ser-A'e t: SEpho e/1.60.;=<a FSE 9a!sG Co tact: Csip:;0100A1<;.16=.0.100:5060D>expires?3600


Mid2 will be on 30 March at the class room Lecture 9-Lecture 16

Signaling (roughly analogous to the switching function described in the last two chapters) protocols (H.323 and Session Initiation Protocol [SIP]) set up the route for the media stream or conversation. Gateway control protocols such as the Media Gateway Control Protocol (MGCP) and MEGACO (also signaling protocols) establish control and status in media and signaling gateways. Routing (using the User Datagram Protocol [UDP] and Transmission Control Protocol [TCP]) and transporting (Real-Time Transport Protocol [RTP]) the media stream (conversation) once the route of the media stream has been established are the functions of routing and transport protocols.

Protocols Related to VoIP

H.323 is comprised of a number of sub-protocols.

It uses protocol H.225.0 for registration, admission, status, call signaling, and control. It also uses protocol H.245 for media description and control, terminal capability exchange, and general control of the logical channel carrying the media stream(s).

Other protocols make up the complete H.323 specification, which presents a protocol stack for H.323 signaling and media transport. H.323 also defines a set of call control, channel setup and codec specifications for transmitting real-time video and voice over networks that dont offer guaranteed service or quality of service (QoS). As a transport, H.323 uses RTP, an Internet Engineering Task Force (IETF) standard designed to handle the requirements of streaming real-time audio and video via the Internet.


A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non-H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and TDM networks

Not included



H.323 Signaling