The Interworking of IP Telephony with Legacy Networks

The Interworking of IP Telephony with Legacy Networks Yang Qiu Valmio 0/ 0080 Helsinki Yang.Qiu@nokia.com Abstract This document describes the Interworking of IP Telephony networks with legacy networks. The legacy networks are, ISDN, GSM and IS-. 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. Introduction 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. H. is an ITU's Standard that specifies Packetbased multimedia communications. It's an umbrella standard that references many other ITU standards. 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. MEGACO is the latest generation call agentgateway signaling protocol and standardized by ITU and IETF. ISUP is a level 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, ISDN, GSM and IS-. The module performing the mapping IP telephony signaling (, H. 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, H. or MGCP/MEGACO. There is typically a Media Gateway (MG) with E/T interfaces (voice from ) and with IP interfaces (VoIP). The MGC and the MG can be merged together in one physical box or kept separate. 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, ISDN, IS-'s gateway, GSM's GMSC/MSC (not include the BSC) as their signaling. ISDN, GSM and IS-are represented by and ISUP is also used for connecting to other networks. R is a very old signaling in the ; in this document there is no intention to take it into consideration. (In UMTS All-IP Core Network, MRF is used for connection between R and MEGACO/H.8) Interworking between and ISUP The first step in initiation of a call-using is to locate a server for the callee. Once a 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 hosts as in a conference call scenario.. -ISUP Gateway In a -ISUP network, should be used to provide ISUP translating across -IP networks interconnections. The -ISUP gateway is used where an IP network (the signaling is ) interfaces with the 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. In nowadays IP networks, should terminate calls directly to an end-user device that are hosted by server or by the. As well, /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 --

will not know whether the receiving (i.e. next -hop) network is a terminating network or a transit network. originator and MGC becomes the terminator. One or more proxies may be used to route the call from the originator to the terminator.. -ISUP Network components The following are the components of a -ISUP network.. : 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.. IP-endpoint: Any sort of device that originates -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:?? 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 and the IP-networks. An MGC communicates ISUP to the and to the IPnetworks and converts between the two.?? -phone: The term used to represent all end-user devices that originate calls.?? Firewalls or edge-elements through which calls may enter the network from that of a peer network.. Proxy: A proxy is a entity that helps route signaling messages to their destinations. Consequently, a proxy might route messages to other proxies (some of which may be co-located with firewalls), MGCs and phones.. The structure of - Network In Figure two LECs (Local Exchange Center) are bridged by the IP network together. 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 interface and sends them out to another ( termination). Let say, a call originates from LEC and be terminated by LEC. The originator is defined as the generator of the setup signaling and the terminator is defined as the consumer of the setup signaling. MGC is thus the SS7/C7 PROXY LEC MGC VoIP Network PROXY MGC LEC Figure : ISUP- inter-connecting SS7/C7 Voice calls do not always have to originate and terminate in the (via MGCs). They may either originate and/or terminate in phones. The alternatives for call origination and termination suggest the following possibilities for calls that transit through an IP network.. to ISUP mapping Figure is the State Machine of the mapping from to ISUP. CAN CEL Not alerting CAN CEL CAN CEL BYE E. CPG CPG Idle Trying Alerting ANM Waiting for Connected CON Figure : - State machine --

.. Call setup 8 9 0 00 TRYING 8X PR 00 OK 8X PR 00 OK 00 OK CPG ANM Figure : - Call Flow. When a user tries to begin a session with a user, the node issues an request.. Upon receipt of an request, the gateway will then map it to an message and then be sent to the ISUP network. The ISUP node indicates that the address is sufficient to set up a call by sending back an message. The 'called party status' code in the message is mapped to a provisional response and returned to the node:?? 80 for 'subscriber free'?? 8 for ' no indication' This response may contain SDP to establish an early media stream (as show in Figure ). If no SDP is 7 presented, the audio will be established in both directions after the ISUP send ANM.. The node sends a PR message to confirm receipt of the provisional response.. The PR 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 provisional response and send it to the node. 9. The node sends a PR message to confirm receipt of the provisional response. 0. The PR is confirmed. Once the user responses, an ANM will be sent to the gateway. Upon receipt of the ANM, the gateway will send a 00 message to the node.. The node, upon receiving an final response (00), will send an to acknowledge... Auto-answer Call setup 00 TRYING 00 CON Figure : - Call Flow (auto-answer). When a user wishes to begin a session with a user, the node issues an request.. Upon receipt of an request, the gateway maps it to an message and sends it to the ISUP network --

. Since the ISUP node is configured for automatic answering, it will send a CON message upon receipt of the. (For the ANSI C7, the message will be an ANM). Upon receipt of the CON/ANM, the gateway will send a 00 message to the node.. The node, upon receiving an final response(00), will send an to acknowledge receipt.. ISUP Setup Failure 00 TRYING XX Figure : - Setup Failure. When a user wishes to begin a session with a user, the node issues an request.. Upon receipt of an request, the gateway maps it to an message and sends it to the ISUP network. Since the ISUP node is unable to complete the call, it will issue a.. The gateway releases the circuit and confirms that it is available for reuse by sending an.. The gate translates the cause code in the to a error response and sends it to the node?? unallocated (ISUP:)? 0 Gone?? no route to network(isup:)? 0 Not found?? no route to destination(isup:)? 0 Not found?? user busy(isup:7)? 8 Busy here?? no user responding(isup:8)? 80 Temporarily unavailable?? no answer from the user (ISUP:9)? 80 Temporarily unavailable?? call rejected(isup:)? 0 Decline?? number changed (ISUP:)? 0 moved Permanently?? Destination out of order(isup:7)? 0 Not found?? Address incomplete(isup:8)? 8 Address incomplete?? Facility rejected(isup:9)? 0 Not implemented?? Normal unspecified(isup:)? 0 Not found. The node sends an to acknowledge receipt of the final response.. Call cancelled by node 7 8 0 00 TRYING 8X PR 00 OK CANCEL 00 OK 87 Figure : - Call Cancelled. When a user wishes to begin a session with a user, the node issues an request.. Upon receipt of an request, the gateway maps it to an message and sends it to the ISUP network. The ISUP node indicates that the address is sufficient to set up a call by sending back an message. The 'called party status' code in the message is mapped to a provisional response and returned to the node: (80 for 'subscriber free', and 8 for ' no indication') This response may contain SDP to establish an early media stream (as show in 9 --

Figure ). If no SDP is present, the audio will be established in both directions after the ISUP send ANM.. The node sends a PR message to confirm receipt of the provisional response.. The PR is confirmed 7. To cancel the call before it is answered, the nodes sends a CANCEL request 8. The CANCEL request is confirmed with a 00 response 9. Upon receipt of the CANCEL request, the gateway sends a message to terminate the ISUP call. 0. The gateway sends a '87 Call Cancelled' message to the node to complete the transaction.. Upon receipt of the message, the remote ISUP node will reply with a message.. Upon receipt of the 87, the node will confirm reception with an.. ISUP to mapping Figure 7 is the State Machine of the mapping from ISUP to. BYE T Progressing 00 8x Idle Trying 8x Alerting 00 Connected 00 xx+ xx+ xx+ Figure 7: - State machine.. Call setup 7 9 0 00 TRYING 8X PR 00 OK 8X PR 00 OK 00 OK CPG ANM Figure 8: - Call Flow. When a user wishes to begin a session with a user, the network generates an message towards the gateway.. Upon receipt of an message, the gateway generates an message, and sends it to an appropriate node based on called number analysis.. By the time an event signifying that the call has sufficient addressing information, the node will generate a provisional response of 80 or greater. It this 80 contains a session description.. Upon receipt of a provisional response of 80 or greater, the gateway will generate an message. If the response is not 80, the will carry a ' called party status' value of ' no indication'. 8. The gateway sends a PR message to confirm receipt of the provisional response.. The PR is confirmed --

7. The node may use further provisional messages to indicate call progress. 8. After an has been sent, all provisional responses will be translated into ISUP CPG message. 9. The gateway sends a PR message to confirm the receipt of the provisional response. 0. The PR is then confirmed. When the node answers the call, it will send a 00 OK message.. Upon receipt of the 00 OK message, the gateway will send an ANM message towards the ISUP node.. The gateway will send an to the node to acknowledge receipt of the final response... Auto-answer Call setup 00 CON.. ISUP Setup Failure XX Figure 0: - Setup Failure. When a user wishes to begin a session with a user, the network generates an message towards the gateway.. Upon receipt of the message, the gateway generates an message, and sends it to an appropriate node based on called number analysis.. The node indicates an error condition by replying with a response with a code of 00 or greater.. The gateway sends an message to acknowledge receipt of the final response.. An ISUP message is generated from the code. Figure 9: - Call Flow (auto-answer). When a user wis hes to begin a session with a user, the network generates an message towards the gateway.. Upon receipt of the message, the gateway generates an message, and sends it to an appropriate node based on called number analysis.. Since the node is set up to automatically answer the call, it will send a 00 OK message.. Upon receipt of the 00 OK message, the gateway will send a CON message towards the ISUP node.. An will sent be the gateway to the node to acknowledge the receipt of the final response. --

.. Call cancelled by node 9 0 8X PR 00 OK CANCEL 00 OK 87 Figure : - Call Cancelled. When a user tries to begin a session with a user, the network generates an message towards the gateway.. Upon receipt of the message, the gateway will then generate an message, and sends it to an appropriate node based on called number analysis.. When an event signifying that the call has sufficient addressing information occurs, the node will generate a provisional response of 80 or greater. 7 8 9. Upon receipt of a message before an final response, the gateway will send a CANCEL towards the node. 0. Upon receipt of the CANCEL, the node will send a 00 response.. The remote node will send a '87 Call Cancelled' to complete the transaction.. The gateway will send an to the node to acknowledge the receipt of the final response.. Normal Release of the connection.. Caller hangs up ( and ISUP initiated) BYE 00 Figure : Initiated For a normal release of the call (reception of BYE), the MGC immediately sends a 00 response. It then releases the resources in the MG and sends an with a cause code of (normal call clearing) to the. Release of resources is confirmed by the with a. In bridging situations, the contained in the BYE is sent to the.. Upon receipt of a provisional response of 80 or greater, the gateway will generate an message with an event code.. The gateway sends a PR message to confirm receipt of the provisional response.. The PR is confirmed BYE 7. If the calling party hanged up before the node answers the call, a message will be generated. 00 8. The gateway frees up the circuit and indicates that it is available for reuse by sending an. Figure : ISUP Initiated -7-

If the release of the connection was caused by the reception of a, the is included in the BYE sent by the MGC... Callee hangs up (Analog ISUP User) Call control H. Multipoint H. Data T.0 For the IP Telephony, we should pay more attention to the H..0 call setup and H. call control standards. 7 8 00 BYE BYE 00 SUS * T Expires * H.:?? Specifies call setup messages which are based on the Q.9?? Specifies gatekeeper messages(registration, Admissions and Status)?? Describes the use of RTP H. specifies a subset of Q.9 messages that can be used by H. implementations. H. follows Q.9'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.- Q.9 will setup a reliable H. channel. Figure : Analog user hangs up In analog, if the callee hanged up in the middle of a call, the local exchange sends a SUS instead of a and starts a timer (T, SUS is network initiated). When the timer expires, the is sent. Interworking Between H. and ISUP. Overview of H. Standards H. is an umbrella standard that can be referred to many other ITU standards as shown in Table. Call setup and control is handle by H..0 and H.. Table : H. standards Network Non- Guaranteed Bandwidth Packetswitched network ( e.g. IP ) Video H., H. G.7, G.7, G.78, G.79 Call Signaling & media packetisation H..0 H.:?? Specifies conference control and capability exchange message?? allows endpoints to specify RTP port numbers and codec types Once the H. channel is setup, the H. message could be connected to control the multimedia session.. The structure of Interworking between H. and To establish a point-to-point H. conference, Two TCP connections are needed. The first of these that must be set up is commonly known as the Q.9 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..0. Once the H. channel has been setup, the Q.9 channel is no longer required. The H. 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. 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 -8-

complete the H. channel can be used to terminate the call. The Figure shows an example of how interconnection of two regular phones connected to the can be accomplished across an IP network. 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.9 transport address of the remote gateway. The Q.9 Setup message will carry the destination telephone number to the remote gateway which can setup a local call across the 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.79 in RTP packets? G.7 in timeslots. CALLING GW VoIP Network Q.9 Setup Contains CALL PARTY NUMBER Q.9 Connect Contains H. Address of GW H. GW Figure : -IP- In the Example of -IP-: CALLED. The gateway GW performs gateway function to map called number to Q.9 transport address of GW. Q.9 Setup carries call party number to GW. Q.9connect enables GW to learn H. transport address of GW. H. has different ways to use the separate H. channel : (these are not in this document area)?? Fast connect?? Tunneled H. message setup After the H. channel between two gateways are setup, they follow the same rule as the normal H. dialog between two terminals. MGCP and MEGACO. Gateway Decomposition Initial VoIP Gateways handles both call signaling and media conversion. The gateway decomposition model removes the call signaling intelligence from the gateway.?? 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). Aims of Gateway Decomposition: Scalability 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. Seamless Integration Many existing Internet telephony solutions require a -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 -stage dialing since the users phone will already be connected to a gateway. SS7 connectivity Existing H. gateways do not support SS7/C7 connectivity and are also unable to support the full set of 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. Gateway decomposition supports fail-over so that if a call agent fails, another call agent can automatically take over.. Gateway Decomposition Architecture TGW: Trunk gateway. -9-

In UMTS All-IP core Network, it is named as the T-SGW Connects to IP network. Performs the media transformation and act according to instructions from call agent. RGW: Residential gateway. In UMTS All-IP core Network, it is named as the R-SGW 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 In UMTS All-IP Core Network, the RGW is used to connect the MAP between UMTS All-IP Core Network and G Legacy network for roaming. Call agent. In UMTS All-IP Core Network, the Call agent is named as ' Call Processing Server' (CPS)?? Call agent controls TGWs and RGWs using MGCP.?? Handles SS7 signaling for trunks that interconnect 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 /H. signaling In the architecture of Figure, if one call agent fails, another one will take over without losing any calls. Call agents terminates SS7 connectivity allow seamless integration with. 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. 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. Fail-over mechanisms are also unable to meet the very low downtime that users have come to expect over the. Gateway decomposition supports fail-over so that if a call agent fails another call agent can automatically take over. MGCP RGW Signal Voice Call agent RGW Internet Call Agent MGCP TGW MGCP SS7GW SS7/ TCAP, SS7/ ISUP SCP STP Figure : Gateway Decomposition Architecture. Call Agent Gateway communication. Call agent asks the gateway to be informed of certain events(e.g. off-hook, on-hook, dialed digits etc). Gateways report events to call agent. Call Agent informs gateway what to do next and what information to be returned.?? Apply tone to endpoint( dialtone, ringing etc)?? Create connection and return IP address and port. MGCP- call flow In the Figure 7, ring is transmitted across the for both parties as this is the most common approach that is used in today's. This is possible because a RGW includes both a standard telephone connection and an IP network connection. -0-

Caller RGW Off-hook Dial tone Digits I:Off-hook Call agent C:Provide dialtone and collect digits I:Digits dialled C:Create connection I: RGW IP address, UDP Port C:Modify Connection to send to TGW IP address, UDP port TGW RTP voice packets SS7 C:Create Connection sending to RGW IP address, UDP port I: TGW IP address, UDP Port Ringing from ANM. MGCP- call flow Caller Off-hook Dial tone Digits Stop Ringing RGW I:Off-hook Call agent C:Provide dialtone and collect digits I:Digits dialled C:Create connection I: Local IP address, UDP Port C:Start Ring C: stop Ringing I: remote IP address, UDP port RTP voice packets agent Invite(IP address, UDP port) 00 Ringing Callee Ringing Ringing Offhook Figure 7: MGCP- call flow I: indicates information C: indicates a command Figure 8: MGCP- call flow 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 residing on an IP network.7 MEGACO Vs /H. MEGACO assumes dumb end points, similar to and IN model. --

/H. assume intelligent end points, similar to Internet model..8 UMTS All-IP Core Network SGSN HSS GPRS HLR UMS IP Network CPS CSCF MGCF GW T-SGW References [] GPP: 'architecture for an ALL IP network', G TR.9 version.0.0 0th June 000 [] GPP: 'Combined GSM and Mobile IP Mobility Handling in UMTS IP CN', TR.9, version.0.0 0th June 000 [] Aparna Vemuri, Jon Peterson: for Telephones (-T)-Context and Architectures, WG, July 000, http://www.softarmor.com/sipwg/teams/sipt/index.ht ml [] Ericsson: Best Current Practice for ISUP to mapping, IETF, September 000, http://www.softarmor.com/sipwg/teams/sipt/index.h tml [] Phillips Omnicom: Voice over IP, Phillips Omnicom, July 000.HERTS SG EL UK [] Srinivas sreemanthula etc: 'RT Hard Handoff Concept for All-IP System, version V.0., and IPMN project. R-SGW BSS MGW MRF Figure 99: UMTS All-IP Core Network It is a Network of Voice over GPRS. and MEGACO are used for their signaling. HSS has parts:?? GPRSHLR?? UMS. UMS will take control of the application level mobility management (serving the CSCF) CPS has parts:?? CSCF. CSCF provides call control service to IP-telephony subscriber.?? MGCF. MGCF control the gateway. GW has 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 R signaling to the signaling over IP. --