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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


Abstract in normal telephony network, ISUP is the 'almighty'
signaling for these networks to connect with each other.
This document describes the Interworking of IP ISUP is used by PSTN, ISDN, IS-41's gateway, GSM's
Telephony networks with legacy networks. The legacy GMSC/MSC (not include the BSC) as their signaling.
networks are PSTN, ISDN, GSM and IS-41. All of these
Legacy networks use the SS7/C7's ISUP for their ISDN, GSM and IS-41are represented by PSTN
gateway. So, in this document we only discuss the and ISUP is also used for connecting to other networks.
Interworking of IP telephony signaling with ISUP.
R2 is a very old signaling in the PSTN; in this
document there is no intention to take it into
1 Introduction consideration. (In UMTS All-IP Core Network, MRF is
used for connection between R2 and MEGACO/H.248)
SIP is an application layer protocol for establishing,
terminating and modifying multimedia sessions. It 's
typically carried over IP. Telephone calls are considered 2 Interworking between SIP and
as a type of multimedia sessions where just audio is ISUP
exchanged.
The first step in initiation of a call-using SIP is to
H.323 is an ITU's Standard that specifies Packet-
locate a SIP server for the callee.
based 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,
MGCP is the 'Media Gateway Control Protocol'. which may be either point-to-point or may be more than
This Protocol defines the communications between 2 hosts as in a conference call scenario.
Media Gateway and Call Agent, so that the Media
Gateway is fully controlled by the call agent.

MEGACO is the latest generation call agent- 2.1 SIP-ISUP Gateway


gateway signaling protocol and standardized by ITU and In a SIP-ISUP network, SIP should be used to
IETF. provide ISUP translating across PSTN-IP networks inter-
ISUP is a level 4 protocol used in SS7/C7 connections.
networks. It typically runs over MTP but it will also run The SIP-ISUP gateway is used where an IP
over IP as well. ISUP is used for controlling telephone network (the signaling is SIP) interfaces with the PSTN
calls in PSTN, ISDN, GSM and IS-41. network (the signaling is ISUP). Such a network may
The module performing the mapping IP telephony frequently be needed to hand a call over to another
signaling (SIP, H.323 and MEGACO) and ISUP is network in order to terminate it. Therefore, such
usually referred to as Media Gateway Controller (MGC). networks do not normally exist in isolation. They have
It has logical interfaces towards both networks, the business relationships with each other, and they are
network carrying ISUP and the network carrying SIP, connected together in order to terminate calls.
H.323 or MGCP/MEGACO. In nowadays IP networks, SIP should terminate
There is typically a Media Gateway (MG) with calls directly to an end-user device that are hosted by SIP
E1/T1 interfaces (voice from PSTN) and with IP server or by the PSTN. As well, SIP/IP networks may
interfaces (VoIP). just serve as a transit network with IP inter-connections
to other networks that have ISUP interfaces. Such a
The MGC and the MG can be merged together in transit network will accept VoIP calls from one network
one physical box or kept separate. and pass them over to another network where they may
be terminated. And, the originating network most often

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will not know whether the receiving (i.e. next -hop) originator and MGC2 becomes the terminator. One or
network is a terminating network or a transit network. more proxies may be used to route the call from the
originator to the terminator.

2.2 SIP-ISUP Network components


The following are the components of a SIP-ISUP VoIP
network. Network PROXY
PROXY
1. PSTN: This is the Public Switched Telephone
Network. It may either refer to the entire inter-
connected collection of local, long-distance
and international phone companies or some MGC1 MGC2
subsets thereof. It could be any kind of SS7/C7 SS7/C7
network, if they use 'MSISDN/MSIN' to locate
their user.
2. IP-endpoint: Any sort of device that originates LEC1 LEC2
SIP-calls to the network may be considered as Figure 1: ISUP-SIP inter-connecting
an IP end-point for the purposes of this
document. Thus, the following devices may Voice calls do not always have to originate and
classify as: terminate in the PSTN (via MGCs). They may either
?? MGC: A Media Gateway Controller originate and/or terminate in SIP phones. The
(MGC) is an entity used to control a alternatives for call origination and termination suggest
gateway (that is typically used to provide the following possibilities for calls that transit through an
conversion between the audio signals IP network.
carried on telephone circuits and data
packets carried over packet networks). The
term MGC is thus used in this document to 2.4 SIP to ISUP mapping
clarify entities that control the point of
inter-connection between the PSTN and the Figure 2 is the State Machine of the mapping from
SIP to ISUP.
IP-networks. An MGC communicates
ISUP to the PSTN and SIP to the IP-
networks and converts between the two. Idle
?? SIP-phone: The term used to represent all
end-user devices that originate SIP calls. INVITE
?? Firewalls or edge-elements through which
calls may enter the network from that of a CAN Trying REL
peer network. CEL E.ACM ACM CON
3. Proxy: A proxy is a SIP entity that helps route
SIP signaling messages to their destinations.
Consequently, a proxy might route SIP Not alerting
messages to other proxies (some of which may CPG
be co-located with firewalls), MGCs and SIP- CAN
phones. CEL
Alerting REL
CPG ANM
2.3 The structure of SIP-PSTN Network CAN
CEL
In Figure 1 two LECs (Local Exchange Center) are Waiting for ACKREL
bridged by the IP network together. SIP is employed as
the VoIP protocol used to set up and tear down VoIP ACK
sessions and calls. The VoIP network receives ISUP
messages over SS7/C7from one PSTN interface and BYE Connected REL
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 Figure 2: SIP-PSTN State machine
setup signaling and the terminator is defined as the
consumer of the SIP setup signaling. MGC1 is thus the

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2.4.1 Call setup presented, the audio will be established in both
directions after the ISUP send ANM.
SIP MGC/MG PSTN 5. The SIP node sends a PRACK message to confirm
receipt of the provisional response.
INVITE
1 6. The PRACK is confirmed
100 TRYING
IAM 7. The ISUP node may issue a variety of CPG
2 messages to indicate, for example, that the call is
Audio being forwarded.
8. Upon receipt of a CPG message, the gateway will
ACM
18X 3 map the event code to a SIP provisional response
4 and send it to the SIP node.
Audio
9. The SIP node sends a PRACK message to confirm
receipt of the provisional response.
PRACK
5 10. The PRACK is confirmed
200 OK
6 11. Once the PSTN user responses, an ANM will be
CPG sent to the gateway
18X 7
8 12. Upon receipt of the ANM, the gateway will send a
PRACK 200 message to the SIP node.
9
13. The SIP node, upon receiving an INVITE final
200 OK
10 response (200), will send an ACK to acknowledge.
ANM
11 2.4.2 Auto-answer Call setup
Conversation
200 OK
12 SIP MGC/MG PSTN
Conversation
INVITE
1
ACK
13 100 TRYING
IAM
2
Figure 3: SIP-PSTN Call Flow Audio

1. When a SIP user tries to begin a session with a CON


PSTN user, the SIP node issues an INVITE request. 3
Conversation
2. Upon receipt of an INVITE request, the gateway 200
will then map it to an IAM message and then be sent 4
to the ISUP network Conversation
3. The ISUP node indicates that the address is
ACK
sufficient to set up a call by sending back an ACM 5
message

4. The 'called party status' code in the ACM message is Figure 4: SIP-PSTN Call Flow (auto-answer)
mapped to a SIP provisional response and returned
1. When a SIP user wishes to begin a session with a
to the SIP node:
PSTN user, the SIP node issues an INVITE request.
?? 180 for 'subscriber free'
2. Upon receipt of an INVITE request, the gateway
?? 183 for ' no indication' maps it to an IAM message and sends it to the ISUP
network
This response may contain SDP to establish an early
media stream (as show in Figure 2). If no SDP is

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3. Since the ISUP node is configured for automatic ?? Destination out of order(ISUP:27)? 404 Not
answering, it will send a CON message upon receipt found
of the IAM. (For the ANSI C7, the message will be ?? Address incomplete(ISUP:28)? 484 Address
an ANM) incomplete
?? Facility rejected(ISUP:29)? 501 Not
4. Upon receipt of the CON/ANM, the gateway will
implemented
send a 200 message to the SIP node.
?? Normal unspecified(ISUP:31)? 404 Not found
5. The SIP node, upon receiving an INVITE final 6. The SIP node sends an ACK to acknowledge receipt
response(200), will send an ACK to acknowledge of the INVITE final response
receipt

2.4.4 Call cancelled by SIP node


2.4.3 ISUP Setup Failure
SIP MGC/MG PSTN
SIP MGC/MG PSTN
INVITE
INVITE 1
1
100 TRYING
100 TRYING IAM
IAM 2
2
Audio
REL
3
ACM
RLC 18X 3
4XX 4 4
5
Audio
ACK
6
PRACK
5
Figure 5: SIP-PSTN Setup Failure 200 OK
6
1. When a SIP user wishes to begin a session with a
PSTN user, the SIP node issues an INVITE request. CANCEL
7
2. Upon receipt of an INVITE request, the gateway 200 OK
maps it to an IAM message and sends it to the ISUP 8 REL
network 9
487
10 RLC
3. Since the ISUP node is unable to complete the call, 11
it will issue a REL. ACK
12
4. The gateway releases the circuit and confirms that it
is available for reuse by sending an RLC. Figure 6: SIP-PSTN Call Cancelled

5. The gate translates the cause code in the REL to a 1. When a SIP user wishes to begin a session with a
SIP error response and sends it to the SIP node PSTN user, the SIP node issues an INVITE request.
?? unallocated (ISUP:1) ? 410 Gone 2. Upon receipt of an INVITE request, the gateway
?? no route to network(ISUP:2)? 404 Not found maps it to an IAM message and sends it to the ISUP
?? no route to destination(ISUP:3)? 404 Not network
found
?? user busy(ISUP:17)? 486 Busy here 3. The ISUP node indicates that the address is
?? no user responding(ISUP:18)? 480 sufficient to set up a call by sending back an ACM
Temporarily unavailable message
?? no answer from the user (ISUP:19)? 480 4. The 'called party status' code in the ACM message is
Temporarily unavailable
mapped to a SIP provisional response and returned
?? call rejected(ISUP:21)? 603 Decline to the SIP node: (180 for 'subscriber free', and 183
?? number changed (ISUP:22)? 301 moved for ' no indication') This response may contain SDP
Permanently to establish an early media stream (as show in

-4-
Figure 2). If no SDP is present, the audio will be 2.5.1 Call setup
established in both directions after the ISUP send
ANM.
SIP MGC/MG PSTN
5. The SIP node sends a PRACK message to confirm
receipt of the provisional response. IAM
1
6. The PRACK is confirmed Audio
INVITE
7. To cancel the call before it is answered, the SIP 2
nodes sends a CANCEL request 100 TRYING
8. The CANCEL request is confirmed with a 200
18X
response 3
9. Upon receipt of the CANCEL request, the gateway Audio
ACM
sends a REL message to terminate the ISUP call. 4
10. The gateway sends a '487 Call Cancelled' message PRACK
5
to the SIP node to complete the INVITE transaction.
200 OK
11. Upon receipt of the REL message, the remote ISUP 6
node will reply with a RLC message. 18X
7 CPG
12. Upon receipt of the 487, the SIP node will confirm 8
reception with an ACK. PRACK
9
200 OK
2.5 ISUP to SIP mapping 10
200 OK
Figure 7 is the State Machine of the mapping from 11
ISUP to SIP.
Conversation
ANM
12
Idle
Conversation
IAM ACK
13

Trying 4xx+ Figure 8: PSTN-SIP Call Flow


REL
T11 18x 200
1. When a PSTN user wishes to begin a session with a
SIP user, the PSTN network generates an IAM
Progressing 4xx+ message towards the gateway.
200 18x 2. Upon receipt of an IAM message, the gateway
REL generates an INVITE message, and sends it to an
appropriate SIP node based on called number
Alerting 4xx+ analysis.
200
REL 3. By the time an event signifying that the call has
sufficient addressing information , the SIP node will
BYE Connected REL generate a provisional response of 180 or greater. It
this 180 contains a session description.
Figure 7: PSTN-SIP State machine
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

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7. The SIP node may use further provisional messages 2.5.3 ISUP Setup Failure
to indicate call progress.
8. After an ACM has been sent, all provisional SIP MGC/MG PSTN
responses will be translated into ISUP CPG
message. IAM
1
9. The gateway sends a PRACK message to confirm Audio
the receipt of the provisional response. INVITE
2
10. The PRACK is then confirmed 4XX
3
11. When the SIP node answers the call, it will send a
ACK
200 OK message. 4 REL
5
12. Upon receipt of the 200 OK message, the gateway
RLC
will send an ANM message towards the ISUP node. 6
13. The gateway will send an ACK to the SIP node to
acknowledge receipt of the INVITE final response. Figure 10: PSTN-SIP Setup Failure

1. When a PSTN user wishes to begin a session with a


2.5.2 Auto-answer Call setup SIP user, the PSTN network generates an IAM
message towards the gateway.
SIP MGC/MG PSTN
2. Upon receipt of the IAM message, the gateway
generates an INVITE message, and sends it to an
IAM
1 appropriate SIP node based on called number
analysis.
Audio
INVITE
2 3. The SIP node indicates an error condition by
replying with a response with a code of 400 or
200
3 greater.
Conversation 4. The gateway sends an ACK message to
CON
4 acknowledge receipt of the INVITE final response.
Conversation 5. An ISUP REL message is generated from the SIP
ACK code.
5

Figure 9: PSTN-SIP Call Flow (auto-answer)

1. When a PSTN user wis hes 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.

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2.5.4 Call cancelled by SIP node
9. Upon receipt of a REL message before an INVITE
SIP MGC/MG PSTN final response, the gateway will send a CANCEL
towards the SIP node.
IAM
1 10. Upon receipt of the CANCEL, the SIP node will
Audio send a 200 response.
INVITE
2 11. The remote SIP node will send a '487 Call
18X Cancelled' to complete the INVITE transaction.
3
12. The gateway will send an ACK to the SIP node to
Audio
ACM acknowledge the receipt of the INVITE final
4 response.
PRACK
5
200 OK
6 2.6 Normal Release of the connection
REL
7 2.6.1 Caller hangs up (SIP and ISUP
RLC initiated)
CANCEL 8
9
200 OK SIP MGC/MG PSTN
10 Conversation
487
11
ACK BYE
12 1
200
Figure 11: PSTN-SIP Call Cancelled 2 REL
3
1. When a PSTN user tries to begin a session with a RLC
4
SIP user, the PSTN network generates an IAM
message towards the gateway.
Figure 12: SIP Initiated
2. Upon receipt of the IAM message, the gateway will
then generate an INVITE message, and sends it to For a normal release of the call (reception of BYE),
an appropriate SIP node based on called number the MGC immediately sends a 200 response. It then
analysis. releases the resources in the MG and sends an REL with
a cause code of 16 (normal call clearing) to the PSTN.
3. When an event signifying that the call has sufficient Release of resources is confirmed by the PSTN with a
addressing information occurs, the SIP node will RLC. In SIP bridging situations, the REL contained in
generate a provisional response of 180 or greater. the BYE is sent to the PSTN.

SIP MGC/MG PSTN


4. Upon receipt of a provisional response of 180 or
greater, the gateway will generate an ACM message Conversation
with an event code.

5. The gateway sends a PRACK message to confirm REL


1
receipt of the provisional response.
RLC
6. The PRACK is confirmed BYE 2
3
7. If the calling party hanged up before the SIP node 200
answers the call, a REL message will be generated. 4

8. The gateway frees up the PSTN circuit and indicates Figure 13: ISUP Initiated
that it is available for reuse by sending an RLC.

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If the release of the connection was caused by the Call control H.245
reception of a REL, the REL is included in the BYE sent
by the MGC.
Multipoint H.323

2.6.2 Callee hangs up (Analog ISUP Data T.120


User)

SIP MGC/MG PSTN For the IP Telephony, we should pay more


Conversation attention to the H.225.0 call setup and H.245 call control
standards.

SUS H.225:
INVITE 1
2 ?? Specifies call setup messages which are based on the
Q.931
200
3 ?? Specifies gatekeeper messages(Registration,
Admissions and Status)
BYE
4 ?? Describes the use of RTP
* T6 Expires * H.225 specifies a subset of Q.931 messages that
REL 5 can be used by H.323 implementations. H.225 follows
Q.931's procedures for circuit mode connection setup.
RLC Although the 'bearer' is actually been signaled for, no
BYE 6
7 actual 'B' channels of the ISDN type exist on the packet
based network. Successful completion of the H.225-
200
8 Q.931 will setup a reliable H.245 channel.

H.245:
Figure 14: Analog user hangs up
?? Specifies conference control and capability
In analog PSTN, if the callee hanged up in the exchange message
middle of a call, the local exchange sends a SUS instead ?? allows endpoints to specify RTP port numbers and
of a REL and starts a timer (T6, SUS is network codec types
initiated). When the timer expires, the REL is sent.
Once the H.245 channel is setup, the H.245
message could be connected to control the multimedia
3 Interworking Between H.323 and session.
ISUP
3.2 The structure of Interworking between
3.1 Overview of H.323 Standards H.323 and PSTN
To establish a point-to-point H.323 conference,
H.323 is an umbrella standard that can be referred
Two TCP connections are needed. The first of these that
to many other ITU standards as shown in Table 1. Call
must be set up is commonly known as the Q.931
setup and control is handle by H.225.0 and H.245.
channel. The caller initiates setup of this TCP
Table 1: H.323 standards connection to a well-known port at the callee. Call setup
messages are then exchanged as defined in H.225.0.
Network Non- Guaranteed
Bandwidth Packet- Once the H.245 channel has been setup, the Q.931
switched network ( e.g. IP ) channel is no longer required. The H.245 channel is then
Video H.261, H.263 used to allow both sides to exchange their audio/video
capabilities and to determine which side will act as the
master.
Audio G.711, G.722, G.728, G.729
Another function carried out over the H.245
Call Signaling & H.225.0 channel is to initiate the setup of RTP sessions for the
media packetisation data transfer and RTCP sessions for delivery monitoring
and feedback reports. Finally when data transfer is

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complete the H.245 channel can be used to terminate the
call.
The Figure 15 shows an example of how 4 MGCP and MEGACO
interconnection of two regular phones connected to the
PSTN can be accomplished across an IP network.
4.1 Gateway Decomposition
The calling party dials the telephone number of the
local gateway followed by the destination telephone Initial VoIP Gateways handles both call signaling
number. The local gateway then maps the destination and media conversion. The gateway decomposition
number to the Q.931 transport address of the remote model removes the call signaling intelligence from the
gateway. The Q.931 Setup message will carry the gateway.
destination telephone number to the remote gateway ?? Gateways are then controlled by external call
which can setup a local call across the PSTN to the agents containing call signaling intelligence
destination telephone thereby completing the end to end ?? Gateways communicate with call agent, uses a
path. Upon completion of call set up, each gateway is specific protocol(e.g. MGCP/MEGACO)
responsible for media conversion in both directions, e.g.
G.729 in RTP packets ? G.711 in timeslots.
4.2 Aims of Gateway Decomposition:
GW1 GW2
Scalability
CALLING VoIP Network CALLED Existing gateways only support a small number of
lines (a few thousand), partly because the gateway must
PSTN PSTN perform full call signaling as well as media conversion.
By removing the intelligence from the gateway and
Q.931 Setup making it a dumb device under the control of a remote
Contains CALL call agent the gateway will be able to support a larger
PARTY NUMBER number of liners.

Q.931 Connect Seamless PSTN Integration


Contains H.245
Many existing Internet telephony solutions require
Address of GW2
a 2-stage dialing where a gateway number must be
dialed prior to dialing the actual destination number.
H.245 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
Figure 15: PSTN-IP- PSTN since the users phone will already be connected to a
gateway.
In the Example of PSTN-IP-PSTN:
SS7 connectivity
1. The gateway GW1 performs gateway function
to map called number to Q.931 transport Existing H.323 gateways do not support SS7/C7
address of GW2 connectivity and are also unable to support the full set of
PSTN services that is accomplished by using SS7/C7.
2. Q.931 Setup carries call party number to GW2
Availability
3. Q.931connect enables GW1 to learn H.245
transport address of GW2 Existing Internet telephony solution have limited
fail-over mechanisms and are also unable to meet the
4. H.323 has 2 different ways to use the separate very low downtime that users have come to expect over
H.245 channel : (these are not in this document the PSTN. Gateway decomposition supports fail-over so
area) that if a call agent fails, another call agent can
?? Fast connect automatically take over.
?? Tunneled H.245 message setup
After the H.245 channel between two gateways are 4.3 Gateway Decomposition Architecture
setup, they follow the same rule as the normal H.245
dialog between two terminals. TGW: Trunk gateway.

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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
MGCP
In UMTS All-IP core Network, it is named as the
R-SGW
SS7GW
For a connection of residential telephone to an IP
network, it's feasible and inexpensive, as created by the RGW SS7/
removal of call intelligence from gateway. The media RGW TGW TCAP,
transformation is performed and the instructions from SS7/
ISUP
call agent are followed accordinglty

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.

In UMTS All-IP Core Network, the Call agent is Signal


named as ' Call Processing Server' (CPS) Voice SCP
?? Call agent controls TGWs and RGWs using
MGCP. Figure 16: Gateway Decomposition Architecture
?? Handles SS7 signaling for trunks that
interconnect PSTN with IP network
?? Interacts with SCPs over SS7 network in 4.4 Call Agent Gateway communication
support of various services ( e.g. routing of
'free-of-charge' telephone numbers to actual 1. Call agent asks the gateway to be informed of
destination) certain events(E.g. off-hook, on-hook, dialed
?? May support SIP/H.323 signaling digits etc)
In the architecture of Figure 16, if one call agent 2. Gateways report events to call agent
fails, another one will take over without losing any calls.
3. Call Agent informs gateway what to do next
Call agents terminates SS7 connectivity allow and what information to be returned.
seamless integration with PSTN.
?? Apply tone to endpoint( dialtone, ringing
Centralized intelligence leads to a rapid etc)
introduction of new services simply by upgrading the ?? Create connection and return IP address
call agent software and making the service available to and port
anyone that will pay for it.
The above architecture will work with existing non- 4.5 MGCP-PSTN call flow
intelligent customer premises equipment (CPE). Also
In the Figure 17, ring is transmitted across the
permits intelligent CPEs to be used that perhaps offer
PSTN for both parties as this is the most common
some additional services that call agent does not support.
approach that is used in today's PSTN. This is possible
Fail-over mechanisms are also unable to meet the because a RGW includes both a standard telephone
very low downtime that users have come to expect over connection and an IP network connection.
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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4.6 MGCP-SIP call flow
Caller Call agent SS7
Caller Call agent Callee
TGW
RGW
RGW SIP agent
Off-hook I:Off-hook
Off-hook I:Off-hook
C:Provide
dialtone and
collect digits C:Provide
dialtone and
collect digits
Dial tone
Digits I:Digits dialled
Dial tone
Digits I:Digits dialled
C:Create
connection
C:Create
connection
I: RGW IP C:Create
address, Connection
UDP Port I: Local IP
sending to RGW IP
address, UDP port address,
UDP Port Invite(IP address,
I: TGW IP UDP port)
C:Modify address,
Connection to send UDP Port Ring-
to TGW IP address, ing
100
UDP port Ringing
IAM C:Start Ring
Ring-
ACM Off-
ing ACM hook
Ringing from PSTN C: stop Ringing I:
remote IP address,
UDP port
ANM
ACM Stop
Ring-
ing RTP voice packets
RTP voice packets

Figure 17: MGCP-PSTN call flow


Figure 18: MGCP-SIP call flow
I: indicates information
I: indicates information
C: indicates a command
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 References
Internet model.
[1] 3GPP: 'architecture for an ALL IP network', 3G TR
23.922 version 1.0.0 30th June 2000
4.8 UMTS All-IP Core Network [2] 3GPP: 'Combined GSM and Mobile IP Mobility
Handling in UMTS IP CN', TR 23.923, version
3.0.0 30th June 2000
HSS [3] Aparna Vemuri, Jon Peterson: SIP for Telephones
(SIP-T)-Context and Architectures, SIP WG, July 14
2000,
GPRS
HLR http://www.softarmor.com/sipwg/teams/sipt/index.ht
CPS ml
[4] Ericsson: Best Current Practice for ISUP to SIP
UMS mapping , IETF, September 2000,
CSCF
http://www.softarmor.com/sipwg/teams/sipt/index.h
tml
MGCF [5] Phillips Omnicom: Voice over IP, Phillips
Omnicom, July 2000.HERTS SG1 1EL UK
[6] Srinivas sreemanthula etc: 'RT Hard Handoff
GW Concept for All-IP System, version V1.0.2, and
SGSN IP Network
T-SGW IPMN project.

R-SGW

BSS MGW

MRF

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.

-12-

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