The Interworking of IP Telephony with Legacy Networks
Yang Qiu
Valmio 10/4
00380 Helsinki
Yang.Qiu@nokia.com
Note: Although many modes of signaling are used
in normal telephony network, ISUP is the 'almighty'
signaling for these networks to connect with each other.
ISUP is used by PSTN, ISDN, IS-41's gateway, GSM's
GMSC/MSC (not include the BSC) as their signaling.
Abstract
This document describes the Interworking of IP
Telephony networks with legacy networks. The legacy
networks are PSTN, ISDN, GSM and IS-41. All of these
Legacy networks use the SS7/C7's ISUP for their
gateway. So, in this document we only discuss the
Interworking of IP telephony signaling with ISUP.
1
ISDN, GSM and IS-41are represented by PSTN
and ISUP is also used for connecting to other networks.
R2 is a very old signaling in the PSTN; in this
document there is no intention to take it into
consideration. (In UMTS All-IP Core Network, MRF is
used for connection between R2 and MEGACO/H.248)
Introduction
SIP is an application layer protocol for establishing,
terminating and modifying multimedia sessions. It 's
typically carried over IP. Telephone calls are considered
as a type of multimedia sessions where just audio is
exchanged.
2
Interworking between SIP and
ISUP
The first step in initiation of a call-using SIP is to
locate a SIP server for the callee.
H.323 is an ITU's Standard that specifies Packetbased multimedia communications. It's an umbrella
standard that references many other ITU standards.
Once a SIP server has been found, the client can
invite the callee to join the communication session,
which may be either point-to-point or may be more than
2 hosts as in a conference call scenario.
MGCP is the 'Media Gateway Control Protocol'.
This Protocol defines the communications between
Media Gateway and Call Agent, so that the Media
Gateway is fully controlled by the call agent.
2.1
MEGACO is the latest generation call agentgateway signaling protocol and standardized by ITU and
IETF.
SIP-ISUP Gateway
In a SIP-ISUP network, SIP should be used to
provide ISUP translating across PSTN-IP networks interconnections.
ISUP is a level 4 protocol used in SS7/C7
networks. It typically runs over MTP but it will also run
over IP as well. ISUP is used for controlling telephone
calls in PSTN, ISDN, GSM and IS-41.
The SIP-ISUP gateway is used where an IP
network (the signaling is SIP) interfaces with the PSTN
network (the signaling is ISUP). Such a network may
frequently be needed to hand a call over to another
network in order to terminate it. Therefore, such
networks do not normally exist in isolation. They have
business relationships with each other, and they are
connected together in order to terminate calls.
The module performing the mapping IP telephony
signaling (SIP, H.323 and MEGACO) and ISUP is
usually referred to as Media Gateway Controller (MGC).
It has logical interfaces towards both networks, the
network carrying ISUP and the network carrying SIP,
H.323 or MGCP/MEGACO.
In nowadays IP networks, SIP should terminate
calls directly to an end-user device that are hosted by SIP
server or by the PSTN. As well, SIP/IP networks may
just serve as a transit network with IP inter-connections
to other networks that have ISUP interfaces. Such a
transit network will accept VoIP calls from one network
and pass them over to another network where they may
be terminated. And, the originating network most often
There is typically a Media Gateway (MG) with
E1/T1 interfaces (voice from PSTN) and with IP
interfaces (VoIP).
The MGC and the MG can be merged together in
one physical box or kept separate.
-1-
will not know whether the receiving (i.e. next-hop)
network is a terminating network or a transit network.
2.2
originator and MGC2 becomes the terminator. One or
more proxies may be used to route the call from the
originator to the terminator.
TypeUnitOrDepartmentHere
TypeYourNameHere
SIP-ISUP Network components
The following are the components of a SIP-ISUP
network.
1.
2.
MGC1

PROXY
MGC2
SS7/C7
SS7/C7
LEC1
IP-endpoint: Any sort of device that originates
SIP-calls to the network may be considered as
an IP end-point for the purposes of this
document. Thus, the following devices may
classify as:

3.
PROXY
PSTN: This is the Public Switched Telephone
Network. It may either refer to the entire interconnected collection of local, long-distance
and international phone companies or some
subsets thereof. It could be any kind of
network, if they use 'MSISDN/MSIN' to locate
their user.

VoIP
Network
LEC2
Figure 1: ISUP-SIP inter-connecting
Voice calls do not always have to originate and
terminate in the PSTN (via MGCs). They may either
originate and/or terminate in SIP phones. The
alternatives for call origination and termination suggest
the following possibilities for calls that transit through an
IP network.
MGC: A Media Gateway Controller
(MGC) is an entity used to control a
gateway (that is typically used to provide
conversion between the audio signals
carried on telephone circuits and data
packets carried over packet networks). The
term MGC is thus used in this document to
clarify entities that control the point of
inter-connection between the PSTN and the
IP-networks. An MGC communicates
ISUP to the PSTN and SIP to the IPnetworks and converts between the two.
SIP-phone: The term used to represent all
end-user devices that originate SIP calls.
Firewalls or edge-elements through which
calls may enter the network from that of a
peer network.
2.4
SIP to ISUP mapping
Figure 2 is the State Machine of the mapping from
SIP to ISUP.
Idle
INVITE
CAN
CEL
Proxy: A proxy is a SIP entity that helps route
SIP signaling messages to their destinations.
Consequently, a proxy might route SIP
messages to other proxies (some of which may
be co-located with firewalls), MGCs and SIPphones.
Trying
E.ACM
ACM
CON
Not alerting
CAN
CEL
CPG
Alerting
2.3
REL
CAN
CEL
The structure of SIP-PSTN Network
In Figure 1 two LECs (Local Exchange Center) are
bridged by the IP network together. SIP is employed as
the VoIP protocol used to set up and tear down VoIP
sessions and calls. The VoIP network receives ISUP
messages over SS7/C7from one PSTN interface and
sends them out to another (PSTN termination). Let say,
a call originates from LEC1 and be terminated by LEC2.
The originator is defined as the generator of the SIP
setup signaling and the terminator is defined as the
consumer of the SIP setup signaling. MGC1 is thus the
CPG
REL
ANM
Waiting for ACK REL
ACK
BYE
Connected
REL
Figure 2: SIP-PSTN State machine
-2-
presented, the audio will be established in both
directions after the ISUP send ANM.
2.4.1 Call setup
SIP
MGC/MG
TypeUnitOrDepartmentHere
The SIP node sends a PRACK message to confirm
TypeYourNameHere
receipt of the provisional response.
PSTN
5.
INVITE
1
100 TRYING
IAM
2
6.
The PRACK is confirmed
7.
The ISUP node may issue a variety of CPG
messages to indicate, for example, that the call is
being forwarded.
8.
Upon receipt of a CPG message, the gateway will
map the event code to a SIP provisional response
and send it to the SIP node.
9.
The SIP node sends a PRACK message to confirm
receipt of the provisional response.
Audio
ACM
18X
4
3
Audio
PRACK
5
10. The PRACK is confirmed
200 OK
6
CPG
18X
8
11. Once the PSTN user responses, an ANM will be
sent to the gateway
7
12. Upon receipt of the ANM, the gateway will send a
200 message to the SIP node.
PRACK
9
13. The SIP node, upon receiving an INVITE final
response (200), will send an ACK to acknowledge.
200 OK
10
ANM
200 OK
12
11
2.4.2 Auto-answer Call setup
Conversation
SIP
Conversation
1
ACK
13
MGC/MG
INVITE
100 TRYING
2.
3.
4.
CON
When a SIP user tries to begin a session with a
PSTN user, the SIP node issues an INVITE request.
Upon receipt of an INVITE request, the gateway
will then map it to an IAM message and then be sent
to the ISUP network
4
5

183 for ' no indication'
200
3
Conversation
ACK
Figure 4: SIP-PSTN Call Flow (auto-answer)
The 'called party status' code in the ACM message is
mapped to a SIP provisional response and returned
to the SIP node:
180 for 'subscriber free'
2
Conversation
The ISUP node indicates that the address is
sufficient to set up a call by sending back an ACM
message

IAM
Audio
Figure 3: SIP-PSTN Call Flow
1.
PSTN
This response may contain SDP to establish an early
media stream (as show in Figure 2). If no SDP is
-3-
1.
When a SIP user wishes to begin a session with a
PSTN user, the SIP node issues an INVITE request.
2.
Upon receipt of an INVITE request, the gateway
maps it to an IAM message and sends it to the ISUP
network
3.

Since the ISUP node is configured for automatic
answering, it will send a CON message upon receipt
of the IAM. (For the ANSI C7, the message will be
an ANM)
4.
Upon receipt of the CON/ANM, the gateway will
send a 200 message to the SIP node.
5.
The SIP node, upon receiving an INVITE final
response(200), will send an ACK to acknowledge
receipt



6.
MGC/MG
SIP
PSTN
1
INVITE
1
100 TRYING
5
IAM
RLC
3
4
4
7
Upon receipt of an INVITE request, the gateway
maps it to an IAM message and sends it to the ISUP
network
Since the ISUP node is unable to complete the call,
it will issue a REL.
4.
The gateway releases the circuit and confirms that it
is available for reuse by sending an RLC.
5.
6
When a SIP user wishes to begin a session with a
PSTN user, the SIP node issues an INVITE request.
3.
8
10
12





18X
ACM
3
PRACK
200 OK
CANCEL
200 OK
487
REL
RLC
ACK
9
11
Figure 6: SIP-PSTN Call Cancelled
The gate translates the cause code in the REL to a
SIP error response and sends it to the SIP node



2
Audio
Figure 5: SIP-PSTN Setup Failure
2.
IAM
Audio
5
1.
PSTN
INVITE
2
ACK
6
MGC/MG
100 TRYING
REL
4XX
The SIP node sends an ACK to acknowledge receipt
of the INVITE final response
2.4.4 Call cancelled by SIP node
2.4.3 ISUP Setup Failure
SIP
Destination out of order(ISUP:27)404 Not
found
Address incomplete(ISUP:28)484 Address
incomplete
Facility rejected(ISUP:29)501 Not
implemented
Normal unspecified(ISUP:31)404 Not found
unallocated (ISUP:1) 410 Gone
no route to network(ISUP:2)404 Not found
no route to destination(ISUP:3)404 Not
found
user busy(ISUP:17)486 Busy here
no user responding(ISUP:18)480
Temporarily unavailable
no answer from the user (ISUP:19)480
Temporarily unavailable
call rejected(ISUP:21)603 Decline
number changed (ISUP:22)301 moved
Permanently
-4-
1.
When a SIP user wishes to begin a session with a
PSTN user, the SIP node issues an INVITE request.
2.
Upon receipt of an INVITE request, the gateway
maps it to an IAM message and sends it to the ISUP
network
3.
The ISUP node indicates that the address is
sufficient to set up a call by sending back an ACM
message
4.
The 'called party status' code in the ACM message is
mapped to a SIP provisional response and returned
to the SIP node: (180 for 'subscriber free', and 183
for ' no indication') This response may contain SDP
to establish an early media stream (as show in
Figure 2). If no SDP is present, the audio will be
established in both directions after the ISUP send
ANM.
TypeUnitOrDepartmentHere
2.5.1 Call setup
SIP
MGC/MG
TypeYourNameHere
5.
The SIP node sends a PRACK message to confirm
PSTN
receipt of the provisional response.
IAM
6.
The PRACK is confirmed
Audio
7.
To cancel the call before it is answered, the SIP
nodes sends a CANCEL request
8.
9.
2
100 TRYING
The CANCEL request is confirmed with a 200
response
3
18X
Audio
Upon receipt of the CANCEL request, the gateway
sends a REL message to terminate the ISUP call.
10. The gateway sends a '487 Call Cancelled' message
to the SIP node to complete the INVITE transaction.
5
6
11. Upon receipt of the REL message, the remote ISUP
node will reply with a RLC message.
7
12. Upon receipt of the 487, the SIP node will confirm
reception with an ACK.
2.5
INVITE
9
10
ISUP to SIP mapping
11
Figure 7 is the State Machine of the mapping from
ISUP to SIP.
T11
18x
Progressing
REL
BYE
4xx+
18X
CPG
8
PRACK
200 OK
200 OK
ACK
ANM
12
Conversation
Figure 8: PSTN-SIP Call Flow
200
1.
When a PSTN user wishes to begin a session with a
SIP user, the PSTN network generates an IAM
message towards the gateway.
2.
Upon receipt of an IAM message, the gateway
generates an INVITE message, and sends it to an
appropriate SIP node based on called number
analysis.
3.
By the time an event signifying that the call has
sufficient addressing information , the SIP node will
generate a provisional response of 180 or greater. It
this 180 contains a session description.
4.
Upon receipt of a provisional response of 180 or
greater, the gateway will generate an ACM message.
If the response is not 180, the ACM will carry a '
called party status' value of ' no indication'.
5.
The gateway sends a PRACK message to confirm
receipt of the provisional response.
6.
The PRACK is confirmed
4xx+
200 18x
Alerting
REL
13
4
200 OK
Conversation
IAM
Trying
ACM
PRACK
Idle
REL
1
4xx+
200
Connected
REL
Figure 7: PSTN-SIP State machine
-5-
TypeUnitOrDepartmentHere
TypeYourNameHere
7.
The SIP node may use further provisional messages
to indicate call progress.
8.
After an ACM has been sent, all provisional
responses will be translated into ISUP CPG
message.
9.
2.5.3 ISUP Setup Failure
SIP
2
10. The PRACK is then confirmed
3
11. When the SIP node answers the call, it will send a
200 OK message.
4
12. Upon receipt of the 200 OK message, the gateway
will send an ANM message towards the ISUP node.
2
3
INVITE
PSTN
ACK
Figure 9: PSTN-SIP Call Flow (auto-answer)
When a PSTN user wishes to begin a session with a
SIP user, the PSTN network generates an IAM
message towards the gateway.
2.
Upon receipt of the IAM message, the gateway
generates an INVITE message, and sends it to an
appropriate SIP node based on called number
analysis.
3.
Since the SIP node is set up to automatically answer
the call, it will send a 200 OK message.
4.
Upon receipt of the 200 OK message, the gateway
will send a CON message towards the ISUP node.
5.
An ACK will sent be the gateway to the SIP node
to acknowledge the receipt of the INVITE final
response.
6
Upon receipt of the IAM message, the gateway
generates an INVITE message, and sends it to an
appropriate SIP node based on called number
analysis.
3.
The SIP node indicates an error condition by
replying with a response with a code of 400 or
greater.
4.
The gateway sends an ACK message to
acknowledge receipt of the INVITE final response.
5.
An ISUP REL message is generated from the SIP
code.
4
1.
5
2.
1
Conversation
5
REL
When a PSTN user wishes to begin a session with a
SIP user, the PSTN network generates an IAM
message towards the gateway.
Audio
CON
ACK
1.
200
Conversation
4XX
Figure 10: PSTN-SIP Setup Failure
2.5.2 Auto-answer Call setup
IAM
INVITE
1
Audio
RLC
13. The gateway will send an ACK to the SIP node to
acknowledge receipt of the INVITE final response.
MGC/MG
PSTN
IAM
The gateway sends a PRACK message to confirm
the receipt of the provisional response.
SIP
MGC/MG
-6-
TypeUnitOrDepartmentHere
TypeYourNameHere
2.5.4 Call cancelled by SIP node
SIP
MGC/MG
IAM
2
3
INVITE
6
ACM
PRACK
10
11
12
10. Upon receipt of the CANCEL, the SIP node will
send a 200 response.
12. The gateway will send an ACK to the SIP node to
acknowledge the receipt of the INVITE final
response.
4
200 OK
CANCEL
Upon receipt of a REL message before an INVITE
final response, the gateway will send a CANCEL
towards the SIP node.
11. The remote SIP node will send a '487 Call
Cancelled' to complete the INVITE transaction.
REL
9
1
Audio
18X
Audio
5
9.
PSTN
RLC
2.6
Normal Release of the connection
TypeUnitOrDepartmentHere
TypeYourNameHere
2.6.1 Caller hangs up (SIP and ISUP
initiated)
7
8
SIP
200 OK
MGC/MG
PSTN
Conversation
487
ACK
When a PSTN user tries to begin a session with a
SIP user, the PSTN network generates an IAM
message towards the gateway.
2.
Upon receipt of the IAM message, the gateway will
then generate an INVITE message, and sends it to
an appropriate SIP node based on called number
analysis.
200
2
Figure 11: PSTN-SIP Call Cancelled
1.
BYE
1
REL
RLC
3
4
Figure 12: SIP Initiated
3.
When an event signifying that the call has sufficient
addressing information occurs, the SIP node will
generate a provisional response of 180 or greater.
4.
Upon receipt of a provisional response of 180 or
greater, the gateway will generate an ACM message
with an event code.
5.
The gateway sends a PRACK message to confirm
receipt of the provisional response.
For a normal release of the call (reception of BYE),
the MGC immediately sends a 200 response. It then
releases the resources in the MG and sends an REL with
a cause code of 16 (normal call clearing) to the PSTN.
Release of resources is confirmed by the PSTN with a
RLC. In SIP bridging situations, the REL contained in
the BYE is sent to the PSTN.
SIP
MGC/MG
PSTN
Conversation
REL
6.
The PRACK is confirmed
3
7.
If the calling party hanged up before the SIP node
answers the call, a REL message will be generated.
4
8.
The gateway frees up the PSTN circuit and indicates
that it is available for reuse by sending an RLC.
BYE
RLC
200
Figure 13: ISUP Initiated
-7-
1
2
If the release of the connection was caused by the
reception of a REL, the REL is included in the BYE sent
by the MGC.
2.6.2 Callee hangs up (Analog ISUP
User)
SIP
MGC/MG
PSTN
SUS
2
RLC
5
6
200
H.245:

In analog PSTN, if the callee hanged up in the
middle of a call, the local exchange sends a SUS instead
of a REL and starts a timer (T6, SUS is network
initiated). When the timer expires, the REL is sent.

3.2
Overview of H.323 Standards
Table 1: H.323 standards
Video
Non- Guaranteed
Bandwidth Packetswitched network ( e.g. IP )
H.261, H.263
Audio
G.711, G.722, G.728, G.729
Call Signaling &
media packetisation
The structure of Interworking between
H.323 and PSTN
To establish a point-to-point H.323 conference,
Two TCP connections are needed. The first of these that
must be set up is commonly known as the Q.931
channel. The caller initiates setup of this TCP
connection to a well-known port at the callee. Call setup
messages are then exchanged as defined in H.225.0.
H.323 is an umbrella standard that can be referred
to many other ITU standards as shown in Table 1. Call
setup and control is handle by H.225.0 and H.245.
Network
Specifies conference control and capability
exchange message
allows endpoints to specify RTP port numbers and
codec types
Once the H.245 channel is setup, the H.245
message could be connected to control the multimedia
session.
Interworking Between H.323 and
ISUP
3.1
Specifies call setup messages which are based on the
Q.931
Specifies gatekeeper messages(Registration,
Admissions and Status)
Describes the use of RTP
H.225 specifies a subset of Q.931 messages that
can be used by H.323 implementations. H.225 follows
Q.931's procedures for circuit mode connection setup.
Although the 'bearer' is actually been signaled for, no
actual 'B' channels of the ISDN type exist on the packet
based network. Successful completion of the H.225Q.931 will setup a reliable H.245 channel.
Figure 14: Analog user hangs up
3
T.120

BYE
8
Data

* T6 Expires *
REL
7
H.323

BYE
4
Multipoint
H.225:
1
200
3
H.245
For the IP Telephony, we should pay more
attention to the H.225.0 call setup and H.245 call control
standards.
Conversation
INVITE
Call control
Once the H.245 channel has been setup, the Q.931
channel is no longer required. The H.245 channel is then
used to allow both sides to exchange their audio/video
capabilities and to determine which side will act as the
master.
Another function carried out over the H.245
channel is to initiate the setup of RTP sessions for the
data transfer and RTCP sessions for delivery monitoring
and feedback reports. Finally when data transfer is
H.225.0
-8-
complete the H.245 channel can be used to terminate the
call.
4
The Figure 15 shows an example of how
interconnection of two regular phones connected to the
PSTN can be accomplished across an IP network.
4.1
The calling party dials the telephone number of the
local gateway followed by the destination telephone
number. The local gateway then maps the destination
number to the Q.931 transport address of the remote
gateway. The Q.931 Setup message will carry the
destination telephone number to the remote gateway
which can setup a local call across the PSTN to the
destination telephone thereby completing the end to end
path. Upon completion of call set up, each gateway is
responsible for media conversion in both directions, e.g.
G.729 in RTP packets  G.711 in timeslots.
GW1
CALLING
PSTN


4.2
CALLED
PSTN
Seamless PSTN Integration
Many existing Internet telephony solutions require
a 2-stage dialing where a gateway number must be
dialed prior to dialing the actual destination number.
This is quite cumbersome for the end-user. However if
gateways are setup as dumb device then they will be
inexpensive enough for residential users to buy and place
in their homes thus avoiding the need for 2-stage dialing
since the users phone will already be connected to a
gateway.
H.245
Figure 15: PSTN-IP- PSTN
In the Example of PSTN-IP-PSTN:
SS7 connectivity
The gateway GW1 performs gateway function
to map called number to Q.931 transport
address of GW2
2.
Q.931 Setup carries call party number to GW2
3.
Q.931connect enables GW1 to learn H.245
transport address of GW2
Existing H.323 gateways do not support SS7/C7
connectivity and are also unable to support the full set of
PSTN services that is accomplished by using SS7/C7.
Availability
Existing Internet telephony solution have limited
fail-over mechanisms and are also unable to meet the
very low downtime that users have come to expect over
the PSTN. Gateway decomposition supports fail-over so
that if a call agent fails, another call agent can
automatically take over.
H.323 has 2 different ways to use the separate
H.245 channel : (these are not in this document
area)


Aims of Gateway Decomposition:
Existing gateways only support a small number of
lines (a few thousand), partly because the gateway must
perform full call signaling as well as media conversion.
By removing the intelligence from the gateway and
making it a dumb device under the control of a remote
call agent the gateway will be able to support a larger
number of liners.
Q.931 Connect
Contains H.245
Address of GW2
4.
Gateways are then controlled by external call
agents containing call signaling intelligence
Gateways communicate with call agent, uses a
specific protocol(e.g. MGCP/MEGACO)
Scalability
Q.931 Setup
Contains CALL
PARTY NUMBER
1.
Gateway Decomposition
Initial VoIP Gateways handles both call signaling
and media conversion. The gateway decomposition
model removes the call signaling intelligence from the
gateway.
GW2
VoIP Network
MGCP and MEGACO
Fast connect
Tunneled H.245 message setup
4.3
After the H.245 channel between two gateways are
setup, they follow the same rule as the normal H.245
dialog between two terminals.
Gateway Decomposition Architecture
TGW: Trunk gateway.
-9-
In UMTS All-IP core Network, it is named as the
T-SGW
Connects PSTN to IP network. Performs the media
transformation and act according to instructions from
call agent.
Call Agent
Call agent
Internet
RGW: Residential gateway.
MGCP
MGCP
In UMTS All-IP core Network, it is named as the
R-SGW
MGCP
SS7GW
For a connection of residential telephone to an IP
network, it's feasible and inexpensive, as created by the
removal of call intelligence from gateway. The media
transformation is performed and the instructions from
call agent are followed accordinglty
RGW
RGW
TGW
SS7/
TCAP,
SS7/
ISUP
In UMTS All-IP Core Network, the RGW is used
to connect the MAP between UMTS All-IP Core
Network and 2G Legacy network for roaming.
STP
PSTN
Call agent.
Signal
In UMTS All-IP Core Network, the Call agent is
named as ' Call Processing Server' (CPS)




SCP
Voice
Call agent controls TGWs and RGWs using
MGCP.
Handles SS7 signaling for trunks that
interconnect PSTN with IP network
Interacts with SCPs over SS7 network in
support of various services ( e.g. routing of
'free-of-charge' telephone numbers to actual
destination)
May support SIP/H.323 signaling
Figure 16: Gateway Decomposition Architecture
Call Agent – Gateway communication
4.4
In the architecture of Figure 16, if one call agent
fails, another one will take over without losing any calls.
Call agents terminates SS7 connectivity allow
seamless integration with PSTN.
1.
Call agent asks the gateway to be informed of
certain events(E.g. off-hook, on-hook, dialed
digits etc)
2.
Gateways report events to call agent
3.
Call Agent informs gateway what to do next
and what information to be returned.

Centralized intelligence leads to a rapid
introduction of new services simply by upgrading the
call agent software and making the service available to
anyone that will pay for it.

4.5
The above architecture will work with existing nonintelligent customer premises equipment (CPE). Also
permits intelligent CPEs to be used that perhaps offer
some additional services that call agent does not support.
Apply tone to endpoint( dialtone, ringing
etc)
Create connection and return IP address
and port
MGCP-PSTN call flow
In the Figure 17, ring is transmitted across the
PSTN for both parties as this is the most common
approach that is used in today's PSTN. This is possible
because a RGW includes both a standard telephone
connection and an IP network connection.
Fail-over mechanisms are also unable to meet the
very low downtime that users have come to expect over
the PSTN. Gateway decomposition supports fail-over so
that if a call agent fails another call agent can
automatically take over.
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Caller
Call agent
4.6
SS7
Caller
TGW
RGW
MGCP-SIP call flow
Call agent
RGW
Off-hook
Callee
SIP agent
I:Off-hook
Off-hook
C:Provide
dialtone and
collect digits
C:Provide
dialtone and
collect digits
Dial tone
I:Digits dialled
Digits
Dial tone
I:Digits dialled
Digits
C:Create
connection
I: RGW IP
address,
UDP Port
C:Modify
Connection to send
to TGW IP address,
UDP port
I:Off-hook
C:Create
connection
C:Create
Connection
sending to RGW IP
address, UDP port
I: Local IP
address,
UDP Port
I: TGW IP
address,
UDP Port
Invite(IP address,
UDP port)
100
Ringing
IAM
C:Start Ring
Ringing
ACM
Ringing from PSTN
ACM
Offhook
C: stop Ringing I:
remote IP address,
UDP port
ANM
ACM
Ringing
Stop
Ringing
RTP voice packets
Figure 17: MGCP-PSTN call flow
RTP voice packets
Figure 18: MGCP-SIP call flow
I: indicates information
C: indicates a command
I: indicates information
C: indicates a command
The diagram above show the call flow for a caller
connected to a RGW and a callee support SIP residing
on an IP network
4.7
MEGACO Vs SIP/H.323
MEGACO assumes dumb end points, similar to
PSTN and IN model.
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SIP/H.323 assume intelligent end points, similar to
Internet model.
4.8
References
[1] 3GPP: 'architecture for an ALL IP network', 3G TR
23.922 version 1.0.0 30th June 2000
[2] 3GPP: 'Combined GSM and Mobile IP Mobility
Handling in UMTS IP CN', TR 23.923, version
3.0.0 30th June 2000
[3] Aparna Vemuri, Jon Peterson: SIP for Telephones
(SIP-T)-Context and Architectures, SIP WG, July 14
2000,
http://www.softarmor.com/sipwg/teams/sipt/index.ht
ml
[4] Ericsson: Best Current Practice for ISUP to SIP
mapping , IETF, September 2000,
http://www.softarmor.com/sipwg/teams/sipt/index.h
tml
[5] Phillips Omnicom: Voice over IP, Phillips
Omnicom, July 2000.HERTS SG1 1EL – UK
[6] Srinivas sreemanthula etc: 'RT Hard Handoff
Concept for All-IP System, version V1.0.2, and
IPMN project.
UMTS All-IP Core Network
Figure 199: UMTS All-IP Core Network
It is a Network of Voice over GPRS. SIP and
MEGACO are used for their signaling.
HSS has 2 parts:


GPRSHLR
UMS. UMS will take control of the application
level mobility management (serving the CSCF)
CPS has 2 parts:


CSCF. CSCF provides call control service to
IP-telephony subscriber.
MGCF. MGCF control the gateway.
GW has 4 parts:



T-SGW, T-SGW converts ISUP to the
signaling over IP.
R-SGW, R-SGW allows roaming from IP
telephony domain to Legacy networks and
Legacy subscriber roaming to IP telephony
domain.
MGW and MRF. MGW+MRF will convert the
voice data and convert the R2 signaling to the
signaling over IP.
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