Академический Документы
Профессиональный Документы
Культура Документы
Yang Qiu
Valmio 10/4
00380 Helsinki
Yang.Qiu@nokia.com
-1-
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-
2.4.1 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
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
-3-
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
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
-5-
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
-6-
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.
8. The gateway frees up the PSTN circuit and indicates Figure 13: ISUP Initiated
that it is available for reuse by sending an RLC.
-7-
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
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
-8-
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.
-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
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
-10-
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
-11-
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
-12-