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10 h.248, Sip, Sigtran and Service Flow of Lmds

The document discusses several signaling protocols: - H.248 protocol is used to control media gateways and implement voice services over packet networks by separating control and media functions. It defines terminations as logical entities that initiate or receive media streams and contexts that describe relationships between terminations. - SIP protocol is introduced for establishing multimedia sessions over IP networks. - SIGTRAN and M3UA/SCTP protocols are discussed for transporting signaling messages over IP. - The document also covers LMSD service flow.
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100% found this document useful (1 vote)
454 views55 pages

10 h.248, Sip, Sigtran and Service Flow of Lmds

The document discusses several signaling protocols: - H.248 protocol is used to control media gateways and implement voice services over packet networks by separating control and media functions. It defines terminations as logical entities that initiate or receive media streams and contexts that describe relationships between terminations. - SIP protocol is introduced for establishing multimedia sessions over IP networks. - SIGTRAN and M3UA/SCTP protocols are discussed for transporting signaling messages over IP. - The document also covers LMSD service flow.
Copyright
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Module Code

Module Name
H.248/SIP/SIGTRAN/M3UA/SCTP
and Service Flow in LMDS
V0603

ZTE University
Mobile-MSS/WDSS Group
ZTE University univ.zte.com.cn
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Course Objectives

Understand Function of
H.248 Protocol
Understand Function of SIP
Protocol
Understand Function of
SIGTRAN/M3UA/SCTP Protocol
Understand LMSD Service Flow

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Agenda

 H.248 Protocol Introduction


 SIP Protocol Introduction
 SigTran and M3UA / SCTP
 Service Flow in LMSD

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What is H.248

Control
MSCe Signaling MSCe
Over IP

H.248
H.248:
H.248 usedininMSCe
:used MSCetotocontrol
controlthe
theMGW
MGWandandimplement
implement
H.248
voice bearing on the packet network. They reflect the idea of
voice bearing on the packet network. They reflect the idea of
separatingcontrol
separating controlfunction
functionfrom
fromthe
themedia
mediaconversion
conversionfunction.
function.
MGW
MGW
Media
MGW
MGW Over IP
MGW

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Bearer Mode of H.248 Protocol

Upper-layer application

H.248

M3UA

SCTP

IP

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Termination

– Termination is a logical entity


on the MG, which initiates/rece MG
MG
ives media stream. Terminatio Cn O2=I1+I3
n is identified with Terminatio Tb
I1
nID. Ta
I2

O3=I1+I2
O1=I2+I3 Tc
I3

Td

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Termination

There are two types of termination:


• Semi-permanent termination
Semi-permanent termination represents the physical entity.
For example, for a TDM channel, as long as this channel
exists in the MG, the termination exists.

• Temporary termination
Temporary termination only exists during the calling
procedure, such as RTP media stream. Once the calling
ends, the termination eliminates.

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Context

– Context means the relationship among multiple


terminations. If context involves more than two
terminations, it describes the topology
(receiver/sender), media blending and/or
switching parameters.

– When a call is originated, the H.248 protocol


can use commands to add terminations
corresponding to calling and called parties in
the context.

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Relationship of H.248
Message/ Transaction / Action /Command

Command1
AH Action 1
Message 1
Command2
Transaction 1
Action 2
Message 2
Transaction 2 

Command n

Action n
Message n

Transaction n

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Commands

• Command is used to operate and manage logical entities


(context and terminations). The protocol defines eight
commands, of which the majority is used for the MGC
(MSCe) to control MG (MGW).

• All of the commands in one action control the same


context.

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H.248 Commands

Add

Modify

Subtract

Move
MSCe MGW
AuditValue

AuditCapabilities

Notify

ServiceChange

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Transport

H.248
MSCe
M3UA/SCTP

IP
H.248
Don’t need three-way
Handshake

Port 2944: Text-encoded


MGW
Port 2945: Binary-encoded

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Agenda

 H.248 Protocol Introduction


 SIP Protocol Introduction
 SigTran and M3UA / SCTP
 Service Flow in LMSD

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SIP, H.323 and H.248

Call Control and Signaling Gateway control Media


Video/
H.323 Audio

H.225
SIP H.248/Megaco
H.245 Q.931 RAS RTP RTCP RTSP

TCP UDP

IP

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What is SIP?

“ SIP: Session Initiation Protocol


• SIP is a multimedia communication protocol established
by IETF. It is a text-based application-layer control
protocol independent of lower-layer protocols, designed to
establish, modify and terminate two-party or multi-party
multimedia sessions over the IP network.


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What is SIP? _Cont.

“ SIP was firstly researched by the MMUSIC IETF workg


roup in 1995 and recommended to be a standard by IETF in
1999.
SIP uses HTTP and SMTP protocols for reference.
SIP is still developing now. Relevant equipment vendors
and service providers have created an SIP forum: www.sipfo
rum.org


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What is SIP? _Cont.

“ SIP supports proxy, redirect and user location


registration functions. It also supports user mobility.
With RTP/RTCP, SDP, RTSP and DNS protocols, it
can support and be applied to voice, video and data
services. In addition, it can be applied to characteristic
services like Presence and Instant Message (similar to
QQ).

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Basic Framework

SIP Network Members

LDAP SIP

LDAP
Location Redirect Registrar
SIP
Server Server SIP Server

SIP SIP

PSTN

User Agent Gateway


Proxy Proxy
Server Server

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Basic Framework
User Agents
– An entity that initiates and terminates sessions. It contains
two functional entities.
– User Agent Clients (UAC) –A functional entity that initi
ates the SIP request.
– User Agent Server (UAS) –A functional entity that recei
ves the SIP request.

– UAC and UAS -Both can terminate a call.

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Basic Framework
Proxy Server
– Associated with redirect server and location server
– Proxy for other clients, with SIP message transfer
and forwarding functions. The message mechanism
of proxy server is similar to that of UAC and UAS.
– Transfer received request messages to other servers
after translating and processing.
– Routing the SIP request and response

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Basic Framework
Location Server
– A database designed to store current location informati
on of the user and used by a SIP REDIRECT or PROX
Y SERVER to obtain information about a called user’s
possible location(s).

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Basic Framework
Redirect Server
– Associated with location server.
– Return the new location information of a user to the call
er. The caller can initiate a new call using the new locati
on information.
– Unlike a proxy server, it does not initiate its own SIP re
quest.
– Unlike a user agent server, it does not accept call termin
ation or terminate a call.

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Basic Framework

Registrar Server

– Accept REGISTER request and accomplish user addres


s registration
– Support authentication

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Basic functions of SIP

 Offer SIP users the location function


Location mode: SIP URL, similar to the E-mail address format

 Performing media format negotiation


Encapsulation protocol: SDP - Session Describle Protocol

 Initiate a session
Request message used: INVITE
 Change a session
Request message used: Re-INVITE (Retransmit INVITE, Cseq
is added)
 Terminate a session
Request messages used: BYE, CANCEL

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SIP message - Request and Response
The message mechanism is based on Client/Server, includes
Request and Response.
• SIP request: • SIP response:
– INVITE- Invite a user to initiate a call. – 1xx- Provisional Response.
– ACK- Used to acknowledge the final response t
– 2xx- Successful Response.
o the INVITE that UA has received, used togeth
er with INVITE. – 3xx- Redirection.
– BYE-USER AGENT uses this method to indica – 4xx- Client Error.
te a call release – 5xx- Server Error.
– CANCEL- Used to cancel a uncompleted reque – 6xx- Global Failure.
st, which does not affect the completed request.
– REGISTER- The client uses this method to regi
ster on the server the addresses in the To field.
– OPTIONS- Used to query the service capabiliti
es.
– INFO- Used to carry outband information, e.g.,
DTMF information.

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• For Invite, it need three-way Handshake: invite—200 ok—
ack.

Bob Laura

(1) INVITE

(2) 180 Ringing

(3) 200 OK

(4) ACK

Session

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SIP Session Setup Process

A session will be set up with SIP in the following 6 steps:


1. Register, locate the user.
2. Media negotiation- Media parameters are generally carried in SDP
mode.
3. The called party will determine whether this call will be accepted.
4. Media streams are established and interact.
5. Call may be changed or processed, e.g., Call forwarding (CF)
6. Call terminated.

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SIP Session Setup and Release
100.1.0.252 100.1.0.25 100.2.0.2 100.2.0.8
Transaction1

User Agent Proxy Server Proxy Server User Agent


1 INVITE
2 100 (Trying) 3 INVITE
4 100 (Trying)
Session Setup 5 INVITE
6 100 (Trying)
9 180 (Ringing) 8 180 (Ringing) 7 180 (Ringing)
Session
12 200 (OK) 11 200 (OK) 10 200 (OK)
13 ACK 14 ACK
15 OPTIONS
Transaction2 16 200 (OK)

RTP MEDIA PATH

18 BYE 17 BYE
Teardown
19 200 (OK) 20 200 (OK)

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Transaction3
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SIP-T(SIP for Telephone)

Basic functions of SIP-T:


 
 Encapsulating ISUP in SIP message bodies

 Mapping ISUP information to SIP message headers


 
 Transfer the signaling in sessions using the INFO request

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SIP Commands VS ISUP Messages

SIP MESSAGE OR RESPONSE ISUP MESSAGE


INVITE IAM or SAM
INFO USR
BYE REL
CANCEL REL
ACK ——
REGISTER ——
18x ACM or CPG
200 OK (to INVITE) ANM or CON
4xx, 5xx, 6xx REL
200 OK (to BYE) RLC

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PSTN switch SIP-T gateway SIP-T gateway PSTN switch
1 IAM
2 INVITE (IAM)
3 IAM
4 100 Trying
5 ACM Transaction1
183 Session
6 180 Progress
Ringing (ACM)
(ACM)
7 ACM
One-way
One-way speech RTP Media One-way speech

session 8 ANM
9 200 OK(ANM)
10 ANM
11 ACK
Two-way speech RTP Media Session Two-way speech

12 REL Transaction2
13 BYE(REL)
14 REL
16 200 OK(RLC) 15 RLC
17 RLC
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Agenda

 H.248 Protocol Introduction


 SIP Protocol Introduction
 SigTran and M3UA / SCTP
 Service Flow in LMSD

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IETF SIGTRAN Protocol Model

SS7 SCCP-User Adaptation Layer V5.2-User Adaptation Layer


SS7 MTP2-User Peer-to-Peer Adaptation Layer
SS7 MTP3-User Adaptation Layer
MAP
SS7 MTP2-User Adaptation Layer

BSSAP TCAP ISDN Q.921-User Adaptation Layer

SCCP H.248 TUP/ISUP Q.931 V5.2

SUA MTP3 MTP3


M3UA IUA V5UA
M2PA M2UA

SCTP

IP
Blue parts are SIGTRAN (Signaling Transport).
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Architecture of the SIGTRAN

The SIGTRAN protocol system consists of two parts, signaling


adaptation layer and signaling transport layer. The lower layer
adopts the standard IP bearer.

SS7 signaling adaptation layer


SIGTRAN
{ Common signaling transfer protocol

Standard IP

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Location of SIGTRAN in the Network

SCN SGW SGW SCN

SIGTRAN
For handover

MSCe MSCe

For H248

MEDIA RTP/RTCP MEDIA

MGW MGW

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SCTP Protocol

• SCTP: Stream Control Transmission Protocol


• A reliable connection-oriented transport protocols
• At the same layer as TCP and UDP in network model
• SCTP is designed to transport PSTN signaling messages over IP
networks

SCTP user SCTP user


application application

SCTP layer SCTP layer

One or several IP addresses


IP IP
Network transport
SCTP end point A SCTP end point B

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What does SCTP offer?

Flow control and Network-level fault


Congestion avoidance tolerance

SCTP
Advantages
Protection against Protection against
flooding attacks

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Why not TCP?

Byte-stream Strict order of


Oriented Transmission control

No application specific
TCP Limited scope of
Control of protocol
parameters Limitations TCP socket

Vulnerable for
DoS attack

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Why not UDP?

Unreliable No acknowledgement

UDP Multi-casting adds


No congestion control
Limitations Unnecessary traffic

… too many

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Development of SCTP

UDP

TCP SCTP USCTP

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SCTP In the SIGTRAN Model

TCAP
Signaling Application
Q931/QSIG MTP3 TUP ISUP SCCP TCAP Layer
Signaling Adaptation
IUA M2UA/M2PA M3UA SUA Layer

Transport Protocol
SCTP

IP IP Transport Layer

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Association

• An association is an SCTP connection.


• The creation of an association is initiated by an SCTP user.
To guarantee the security and prevent malicious attacks,
the cookie mechanism is adopted during the starting of an
association.
• The connection creation and release is used for the
transition of the connection status and abnormality
handling.

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M3UA Protocol

• M3UA provides powerful routing function, perfect


signaling network management function and powerful
networking capability and is applicable to inter-network
telephone interconnection.

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M3UA Application in SGW

SEP STP SG(SGP) MSCe(ASP)

ISUP ISUP NIF ISUP

MTP3 MTP3 M3UA M3UA

MTP2 MTP2 SCTP SCTP

MTP1 MTP1 IP IP

TDM IP

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M3UA Application between IPSPs

IP
IPSP IPSP

SCCP USER SCCP USER

SCCP SCCP

M3UA M3UA

SCTP SCTP

IP IP

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Conception of M3UA

• Application Server (AS)


– A logic entity that serves specific routing keys. It includes a group of ap
plication server processes, one or some of which are activated to handle
services.
• Application Server Process (ASP)
– It is the process instance of the AS and is used as the active or standby p
rocess of the AS. ASP can include SCTP end point and can be configur
ed to handle several AS signaling services.

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Conception of M3UA.

• IP Server Process (IPSP)


– It is the process instance based on IP application. IPSP is
essentially identical with ASP, but uses point-to-point M3UA
instead of SG service.
• Signaling Gateway (SG)
– SG receives or sends the SS7 upper-layer user messages at the
boundary between IP network and SS7 signaling network.
– It is a signaling point in SS7 signaling network. It includes one or
several signaling gateway processes, one or more of which
normally handles the service.
• Signaling Gateway Process (SGP)
– SGP is the process instance of signaling gateway and is used as the
active, standby or load sharing process of the signaling gateway.

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Conception of M3UA.

• Routing Key
– A routing key describes a set of SS7 signaling parameters and
parameter values and uniquely defines a signaling service handled by
a specific application server. The parameters of the routing keys can
not be based on several destination signaling point codes.
– The routing keys used in M3UA include: DPC, SIO+DPC,
SIO+DPC+OPC and SIO+DPC+OPC+CIC.
• Routing context
– It uniquely identifies the values of the routing keys.

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Agenda

 H.248 Protocol Introduction


 SIP Protocol Introduction
 SigTran and M3UA / SCTP
 Service Flow in LMSD

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3G Call 3G Service Flow
3G BS OMSCe OMGW HLRe SMSCe SMGW 3G BS
CM Service Request OBSC-Context
Megaco: ADD [RTP, sendrecv, SDP-BS1]; ADD [RTP, recvonly] T0
Megaco: Reply [T1, T2]
Assignment Request OMGW-Context
Assignment Complete
ANSI-41: LOCREQ [DGTSDIAL] T1
ANSI-41:ROUTREQ [MSID] T2
ANSI-41:routreq [TLDN]
ANSI-41: locreq [TRMLIST, REDIND]

SIP-T/SIP: INVITE [IAM, T2]

Paging Request

SIP-T/SIP: 183 Session Progress[ACM] /180 Ringing[ACM]

Megaco:MODIFY[Ringback]
Megaco: Reply []
Origination-Side Ringback

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3G Call 3G Service Flow _Cont.
3G BS OMSCe OMGW HLRe SMSCe SMGW 3G BS
OBSC-Context
Paging Response
Megaco: ADD [RTP, Ringback, sendrecv, T2]; ADD [RTP, sendrecv, SDP-BS2] T0
Megaco: Reply [T3, T4]

OMGW-Context
SIP-T/SIP: 183 Session Progress [T3]
T1
Megaco:MODIFY[Ringback off];MODIFY[RTP,T3,sendrecv]; T2
Megaco: Reply []
Bearer Update Request
Bearer Update Response
SMGW-Context
Termination-Side Ringback
Assignment Request T3
Assignment Complete T4
Connect
SIP-T/SIP: 200 OK(INVITE) [ANM]
Megaco: Modify [Ringback off] SBSC-Context

Megaco: Reply T5
SIP-T/SIP: ACK (INVITE) []
Conversation
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Handover from 3G BS TO 3G BS
Before Handoff After Handoff
39 39
Terminating/ Terminating/ Terminating/ Terminating/
Anchor MSCe Anchor MGW Anchor MSCe Anchor MGW
yy
zz zz yy
yy
zz

39 39 39
Originating/ Originating/ Originating/ Originating/ Target/ Target/
Anchor MSCe Anchor MGW Anchor MSCe Anchor MGW Anchor MSCe Anchor MGW

A1p A2p A1p A2p

Originating Originating Target


BS BS BS

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Service Flow of Handover
from 3G BS TO 3G BS
3G ABS AMSCe AMGW TMGW TMSCe 3G TBS ABSC-Context
HANDOFF REQUIRED Ta
H.248: ADD[C1, $, isolate]
H.248: Reply[Td]
ANSI41: FACDIR2[vCIC] AMGW-Context

SIP: INVITE [ SDP-Td; vCIC ] Tb Tc


H.248: ADD[$, $, $, sendreceive]
H.248: Reply[C2, Te, Tf] Td
HANDOFF REQUEST [Tf]
HANDOFF REQUEST ACK [Tg] TMGW-Context
H.248: MODIFY[C2, Tf, Tg]
H.248: Reply[] Te Tf
SIP: 200 OK (INVITE) [SDP-Te]
H.248: MODIFY[C1, Td, Te]
H.248: Reply[]
SIP: ACK TBSC-Context

ANSI41: facdir2 Tg
ANDOFF COMMAND
HANDOFF COMMENCED
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Service Flow of Handover
from 3G BS TO 3G BS _Cont.
3G ABS AMSCe AMGW TMGW TMSCe 3G TBS ABSC-Context
HANDOFF COMPLETE Ta
ANSI41: MSONCH
H.248: MODIFY[C1, Tb, isolate, Td, bothway]
H.248: Reply[] AMGW-Context

H.248: Subtract[Tb] Tb Tc
H.248: Reply[]
CLEAR COMMAND Td
CLEAR COMPLETE
BEARER UPDATE REQUEST TMGW-Context
BEARER UPDATE RESPONSE
Te Tf

TBSC-Context

Tg

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Thanks !
Question is welcome

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