1. Public Switched Telephone Networks vs. Internet Protocol Networks

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1 Internet Protocol (IP)/Intelligent Network (IN) Integration Tutorial Definition Internet telephony switches enable voice calls between the public switched telephone network (PSTN) and Internet protocol (IP) networks. However, because they lack signaling system 7 (SS7) transaction capabilities, Internet telephony switches cannot access PSTN databases which support intelligent network (IN) services, such as local number portability (LNP) and toll-free (800) databases. The integration of Internet protocol and intelligent networks is an important step in the convergence of voice and data networks. Tutorial Overview This tutorial suggests a possible evolution from existing Internet telephony switch (ITS) technology to an SS7-enabled Internet telephony service switching point (IT-SSP) with intelligent network service capabilities. An overview of common terminology and standards is provided along with examples of enhanced service call flows. Topics 1. Public Switched Telephone Network vs. Internet Protocol Networks 2. The Internet Telephony Switch 3. The Internet Telephony Signaling Point 4. The IP to SS7 Network Gateway 5. The Internet Telephony Service Switching Point 6. Future of Internet Telephony in the PSTN 7. Self-Test 8. Acronym Guide 9. Related Products and Services (Web TecPreviews) 1. Public Switched Telephone Networks vs. Internet Protocol Networks The public switched telephone network provides users with dedicated, end-to-end circuit connections for the duration of each call. Circuits are reserved between the originating switch, tandem switches (if any), and the terminating switch based on the called party number. The PSTN also provides access to intelligent network services using the SS7 protocol. SS7 is used for basic call setup, management, and tear down and to query databases that support intelligent network services such as local number portability, mobile subscriber authentication and roaming, virtual private networking, and toll-free (800) service. Unlike the circuit-switched PSTN, packet-switched IP networks provide shared, virtual circuit connections between users. Bandwidth is dynamically allocated for improved utilization of network capacity. IP packets are routed to the destination IP address contained within the header of each packet. Packets may travel over separate network paths before arriving at their final destination for reassembly and resequencing. The transmission speed between any two users can change dramatically based on the dynamic number of users sharing the common transmission medium, their bandwidth requirements, the capacity of the transmission medium, and the efficiency of the network routing and design. 1 of 13

2 Voice and Video over IP In November 1996, the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T) ratified the H.323 specification that defines how delay-sensitive voice and video traffic is transported over local area networks (LANs). The ITU-T warns that operating H.323 terminals over multiple LAN segments (such as the Internet) may result in poor performance due to the lack of quality-of-service (QoS) guarantees. Until new IP signaling protocols, such as the resource reservation protocol (RSVP), are ratified and implemented in network routers, it is not possible to reserve end-to-end connections over IP networks to guarantee a quality of service equivalent to the PSTN. In the interim, vendors are selling H.323 compatible devices primarily for use over private intranets (rather than the public Internet) where quality-of-service can be controlled. In the Internet, congestion resulting from inadequate bandwidth leads to long delays in the delivery of packets. For voice data, packets that are lost or discarded result in gaps, silence, and clipping in the audio playback. H.323 gateways interface IP networks to the PSTN. Gateways digitize and compress voice calls from the PSTN into IP packets for routing to another gateway (for forwarding to the PSTN) or to an H.323 terminal, such as a PC equipped with H.323 software, a microphone, and speakers. H.323 terminals support the encoding/decoding and packetization/sequencing of information exchanged with other H.323 terminals or gateways. Figure 1 shows the interconnection of the PSTN to H.323 gateways and H.323 terminals. Figure 1: H.323 Gateways/Terminals to PSTN Interconnection Gateways transmit call signaling information to the PSTN using multifrequency (MF) tones, ISDN D-channel signaling, or other forms of signaling. The gateway may also provide enhanced call-processing capabilities such as call prompting and voice recognition. 2. The Internet Telephony Switch The first generation of internet telephony switch (ITS) achieves a limited integration between the PSTN and IP networks. Calls between the PSTN and IP network are supported but intelligent network capabilities, such as toll-free calling and number portability, are unavailable because of the lack of SS7 support in the ITS. The ITS consists of both gateways and gatekeepers (see Figure 2). Figure 2: Internet Telephony Switch 2 of 13 8/26/98 2:51 PM

3 Gatekeepers use the signaling information provided by H.323 gateways or H.323 terminals to provide directory services. The gatekeeper determines the communication path between the originating and terminating gateways and the PSTN using originating/terminating IP addresses and/or called/calling telephone numbers. To provide PSTN/IP routing, gatekeepers maintain a database of remote gatekeeper IP addresses and associated PSTN dialing plan routing information. PSTN to IP to PSTN Call Flow The following example traces a sample call flow which originates in the PSTN, traverses an IP network, and terminates back in the PSTN (see Figure 3): Figure 3: PSTN to IP to PSTN Call Flow 1. An Internet telephony service subscriber dials an access number provided by the Internet telephony service provider. 2. The call is routed by the PSTN to the "access" Internet telephony switch. 3. The gateway plays an announcement requesting that the subscriber enter the destination telephone number to be called. The destination digit information is sent to the gatekeeper. 4. The gatekeeper determines a destination gatekeeper IP address based on the destination digit information. An IP packet requesting the availability status of the destination gateway is sent to the destination gatekeeper. 5. The destination gatekeeper responds to the request by providing destination gateway availability and IP address information. The originating gatekeeper then transfers this information to the originating gateway. 6. The originating gateway sets up a virtual circuit to the destination gateway. This circuit is identified by a call reference variable (CRV) that will be used by both gateways for the duration of the call to identify all IP packets associated with this particular call. 7. The destination gateway selects an outgoing PSTN voice trunk and signals to the PSTN switch to attempt to setup a call to the dialed telephone number. 8. If the PSTN switch signals that the call setup is successful and the called party has answered, IP 3 of 13 8/26/98 2:51 PM

4 signaling messages are relayed to the originating gatekeeper. The gatekeeper then signals the originating gateway, which in turn signals the originating PSTN switch to indicate that the call is now completed. In the IP network, voice packets are exchanged between the gateways for the duration of the call. H.323 Terminal to PSTN Call Flow The following example traces a sample call flow which originates from an IP-based H.323 terminal (e.g., a personal computer) and terminates in the PSTN (see Figure 4): Figure 4: H.323 Terminal to PSTN Call Flow 1. An Internet telephony subscriber initiates a Internet telephony call setup procedure at an H.323 terminal (e.g., a PC). The H.323 terminal sends an IP packet specifying the destination digits information to a preselected gatekeeper. 2. The call scenario from this point onward is similar to steps four through eight described in the PSTN to IP to PSTN call flow (replace "originating gateway" with "originating terminal"). PSTN to H.323 Terminal Call Flow A sample call flow that starts from the PSTN and terminates at an IP-based H.323 terminal is described in Figure 5: Figure 5: PSTN to H.323 Terminal Call Flow 1. An Internet telephony subscriber dials an access number that has provided by the Internet telephony service provider. 2. The call is routed by the PSTN to the "access" Internet telephony switch. 3. The gateway plays an announcement requesting that the subscriber enter the destination number to be called. (In this case, the telephone number is assigned to an H.323 terminal.) The collected destination digit information is sent in a call setup request message to the gatekeeper. 4. The gatekeeper determines a destination gatekeeper IP address based on the destination digit information. An IP packet requesting the availability status for the destination H.323 terminal is sent to the destination gatekeeper. 4 of 13 8/26/98 2:51 PM

5 5. The destination gatekeeper responds to the request by providing the destination terminal availability status and IP address information to the originating gateway. 6. The originating gateway sets up a virtual circuit to the destination H.323 terminal. This circuit is identified by a call-reference variable that will be used by both the originating gateway and the H.323 terminal for the duration of the call to identify all IP packets associated with this particular call. 7. If the H.323 terminal indicates that call setup is successful and the called party has answered, IP signaling messages are sent to the originating gatekeeper, which then signals the originating gateway. The originating gateway signals the originating PSTN switch to indicate that the call is now completed. The exchange of IP packets proceeds until either the calling or called party terminates the call. 3. The Internet Telephony Signaling Point To manage voice circuits in the PSTN, an H.323 gateway may support the SS7 ISDN user part (ISUP). ISUP defines the protocol and procedures used to set-up, manage, and release voice circuits between terminating line exchanges (i.e., between a calling party and a called party) in the PSTN. ISUP is used for both ISDN and non-isdn calls (see Figure 6). Figure 6: Internet Telephony Signaling Point SS7 messages are exchanged between the H.323 gateway and the PSTN switch (called a service switching point [SSP] in SS7 parlance) over dedicated channels called signaling links. The signaling link may be provisioned as an E-1 (2048 kbps; kbps channels), DS-1 (1544 kbps; 24 64kbps channels), V.35 (64 kbps), DS-0 (64 kbps), and DS-0A (56 kbps) connection. The SSP can originate, terminate, or tandem calls to other switches in the PSTN. Because the SS7 network is critical to call processing, signaling links are usually provisioned in pairs to ensure service availability in the event of an isolated failure. Traffic is shared across all links in the linkset. If one of the links fails, signaling traffic is rerouted over another link in the linkset. If an error occurs on a signaling link (e.g., a message is corrupted or arrives out of sequence), the message (or set of messages) is retransmitted. The primary disadvantage of this architecture is that most H.323 gateways do not currently support a sufficient number of SS7 voice trunks to justify the cost of one (or more) SS7 link(s) to the PSTN. For example, a redundant pair of 64 kbps SS7 links can support the signaling requirements of several thousand voice trunks. 5 of 13 8/26/98 2:51 PM

6 4. The IP to SS7 Network Gateway The cost of providing SS7 signaling services to an Internet telephony switch can be reduced by incorporating an SS7-to-IP gateway. The SS7-to-IP gateway (not to be confused with an H.323 gateway) provides cost-effective SS7 interconnection by concentrating signaling traffic across multiple H.323 gateways onto fewer SS7 links (see Figure 7). Figure 7: SS7-to-IP Gateway The SS7-to-IP gateway provides ISUP signaling services to each H.323 gateway. The server also provides an interface to allow the exchange of signaling information with each H.323 gateway. 5. Internet Telephony Service Switching Point To access IN databases, SS7 transaction capabilities application part (TCAP) support can be added into the SS7-to-IP gateway to create an Internet telephony service switching point (IT-SSP). The IT-SSP consists of gateways, gatekeepers, and an SS7-to-IP gateway with SS7 ISUP and TCAP support (see Figure 8). Figure 8: Internet Telephone Service Switching Point 6 of 13 8/26/98 2:51 PM

7 The SS7-to-IP gateway launches TCAP queries into the PSTN to access IN databases (service control points) For example, TCAP queries are used to fetch routing number(s) associated with a dialed 800/888 number, to access local number portability databases, and to verify the personal identification number (PIN) of a calling card user. By supporting both ISUP and TCAP traffic, SS7-to-IP gateways allow H.323 terminal users to gain access to the full suite of IN services available in the PSTN. The SS7-to-IP gateway connects to an SS7 packet switch called a signal transfer point (STP). An STP routes each incoming message to an outgoing signaling link based on routing information contained in the SS7 message. Because it acts as a network hub, an STP provides improved utilization of the SS7 network by eliminating the need for direct links between signaling points. An STP may perform global title translation, a procedure by which the destination signaling point is determined from digits present in the signaling message (e.g., the dialed 800 number, calling card number, or mobile subscriber identification number). An STP can also act as a firewall to screen SS7 messages exchanged with other networks. H.323 Terminal to PSTN Call Flow with IN Service Access Figure 9 illustrates a sample call flow that originates from an H.323 terminal, accesses a PSTN IN local number portability (LNP) database, and terminates in the PSTN. Note that to complete this call, both TCAP (to query the database and process the response) and ISUP (to setup, manage, and release the call in the PSTN) are required. Figure 9: H.323 Terminal to PSTN Call Flow with IN Service Access 1. An Internet telephony service subscriber initiates an Internet telephony call setup procedure at an H.323 terminal (e.g., a PC). The H.323 terminal sends an IP packet specifying the destination digits information to a pre-selected originating gatekeeper. 2. The originating gatekeeper determines that the destination digits are in a ported exchange (NPA-NXX) and that an LNP SCP must be queried to determine whether the dialed number has been ported to another PSTN switch. 3. The originating gatekeeper sends an LNP query request to the SS7-to-IP gateway. The SS7-to-IP gateway encodes and sends an SS7 LNP TCAP query message containing the original dialed number over an SS7 signaling link to the signal transfer point, which routes the message to the LNP SCP. 4. Upon receiving an SS7 LNP TCAP response message from the SCP (via the STP), the SS7-to-IP gateway sends a formatted response to the originating gatekeeper in an IP packet. 5. The originating gatekeeper determines the destination gatekeeper IP address based on the information contained in the LNP response message and sends a request for destination gateway availability status to the destination gatekeeper. 7 of 13 8/26/98 2:51 PM

8 6. The destination gatekeeper responds to the request by sending destination gateway availability status and associated IP address information to the originating gatekeeper which then sends the message to the H.323 terminal. 7. The H.323 terminal selects a destination gateway and sets up a virtual circuit. This circuit is identified by a call reference variable that will be used by both the H.323 terminal and the destination gateway for the duration of the call to identify all IP packets associated with this particular call. 8. The destination gateway selects an outgoing PSTN voice trunk and sends this information in an IP packet to the destination SS7-to-IP gateway. The SS7-to-IP gateway encodes an SS7 ISUP message and sends it to the PSTN switch (via the STP) in an attempt to set up a call to the dialed telephone number. 9. If the PSTN switch, using SS7 ISUP messaging, signals to indicate the call setup is successful and the called party has answered, the SS7-to-IP gateway generates a corresponding signaling message packet and sends it to the originating gatekeeper. The gatekeeper then signals the originating H.323 terminal to indicate the call is now completed. The originating gateway signals the originating PSTN switch to indicate that the call is now completed. The exchange of IP packets proceeds until either the calling or called party terminates the call. 6. Future of Internet Telephony in the PSTN Internet telephony switches need to support SS7 before IT switches can become widely usable. When new IP signaling protocols, such as the resource reservation protocol (RSVP), are standardized and implemented in network routers, the quality of voice over the Internet (VoN) will improve to levels satisfactory to most users. In the interim, voice over IP (VoIP) technology will be used by large customers in their private intranets and extranets (rather than over the public Internet) where quality-of-service can be controlled. The convergence of voice and data networks means that the venerable PSTN will benefit from rapid innovations in Internet protocol technology over the next ten years. New affordable services, such as videoconferencing, promise to herald in a new era of innovative, cost-effective services. 7. Self-Test 1. SS7 is required to setup a call from an H.323 terminal to a PSTN telephone. True or False? 2. The SS7 protocol enables an Internet telephony service switching point to support. a. toll free calling b. calling card calling c. virtual private network services d. number portability e. all of the above 3. Each H.323 gateway in an Internet telephony service switching point needs its own dedicated 8 of 13 8/26/98 2:51 PM

9 SS7 link. True or False? 4. The ITU H.323 specification defines. a. how to set up and tear down a phone call b. how to route a call on the Internet c. how voice, video and data traffic will be transported over local area networks d. a compression scheme for voice traffic e. how an Internet server should connect to the PSTN 5. Each packet associated with a particular call on the IP network is identified by. a. the dialed digits b. the originating gatekeeper's IP address c. a call reference variable (CRV) d. a point code e. none of the above 6. A toll-free call from an H.323 terminal to the PSTN requires the use of. a. ISUP messages b. TCAP messages c. neither ISUP nor TCAP messages d. both ISUP and TCAP messages 7. IP networks provide guaranteed bandwidth for a each phone call. True or False? 8. Within an Internet telephony service switching point, the mapping of telephone numbers to IP addresses is maintained by the gatekeepers. True or False? 9. H.323 terminals can initiate as well as receive calls. True or False? 10. SS7 enables Internet telephony switches to provide enhanced call processing capabilities, such as call prompting and voice recognition. True or False? 11. Internet telephony service switching points can support calls. 9 of 13 8/26/98 2:51 PM

10 a. from the PSTN to the IP network b. from the PSTN to the PSTN via the IP network c. from the IP network to the PSTN d. from the IP network to the IP network e. all of the above Check score Clear You scored 0 percent correct. Correct answers Now it's your turn to give us feedback. On a scale of 1 to 5, where 1 is the lowest and 5 is the highest, rate this tutorial in terms of the following factors: Clarity Completeness Accuracy Overall 1 Lowest 1 Lowest 1 Lowest 1 Lowest Highest 5 Highest 5 Highest 5 Highest Other comments Your (optional) Would you like to be notified about future tutorials? Send in feedback Clear form Correct Answers 1. SS7 is required to setup a call from an H.323 terminal to a PSTN telephone. True or False? 2. The SS7 protocol enables an Internet telephony service switching point to support. 10 of 13 8/26/98 2:51 PM

11 a. toll free calling b. calling card calling c. virtual private network services d. number portability e. all of the above 3. Each H.323 gateway in an Internet telephony service switching point needs its own dedicated SS7 link. True or False? 4. The ITU H.323 specification defines. a. how to set up and tear down a phone call b. how to route a call on the Internet c. how voice, video and data traffic will be transported over local area networks d. a compression scheme for voice traffic e. how an Internet server should connect to the PSTN 5. Each packet associated with a particular call on the IP network is identified by. a. the dialed digits b. the originating gatekeeper's IP address c. a call reference variable (CRV) d. a point code e. none of the above 6. A toll-free call from an H.323 terminal to the PSTN requires the use of. a. ISUP messages b. TCAP messages c. neither ISUP nor TCAP messages d. both ISUP and TCAP messages 7. IP networks provide guaranteed bandwidth for a each phone call. True or False? 8. Within an Internet telephony service switching point, the mapping of telephone numbers to IP addresses is maintained by the gatekeepers. True or False? 9. H.323 terminals can initiate as well as receive calls. True or False? 10. SS7 enables Internet telephony switches to provide enhanced call processing capabilities, such as call prompting and voice recognition. True or False? 11. Internet telephony service switching points can support calls. 11 of 13 8/26/98 2:51 PM

12 a. from the PSTN to the IP network b. from the PSTN to the PSTN via the IP network c. from the IP network to the PSTN d. from the IP network to the IP network e. all of the above 8. Acronym Guide ATM CRV DSP asynchronous transfer mode call reference variable digital signal processing H.323 a specification defining how voice, video, and data traffic will be transported on the Internet IP ISDN ISUP ITS IT-SP IT-SSP ITU LAN LNP MF MTP Internet protocol integrated services digital network ISDN user part ISDN user part (ISUP) messages are connection-oriented messages used to set up and tear down telephone calls. ISUP defines a handshaking protocol that initiates the phone call, reserves a path for the voice or data between the originating and destination devices, and ultimately releases the call. Internet telephone switch Internet telephone switching point Internet telephone service switching point International Telecommunications Union local area network local number portability multiple frequency message transfer part MTP Level 1 The lowest level of the SS7 protocol, MTP Level 1, is equivalent to the OSI physical layer. MTP Level 1 defines the physical, electrical, and functional characteristics of the digital signaling link. Physical interfaces defined include E-1 (2048 kbps; kbps channels), DS-1 (1544 kbps; kbps channels), V.35 (64 kbps), DS-0 (64 kbps), and DS-0A (56 kbps). MTP Level 2 MTP Level 2 ensures accurate end-to-end transmission of a message across a signaling link. Level 2 implements flow control, message sequence validation, and error checking. When an error occurs on a signaling link, the message (or set of messages) is retransmitted. MTP Level 2 is equivalent to the OSI data link layer. MTP Level 3 MTP Level 3 provides message routing between signaling points in the SS7 network. MTP Level 3 is equivalent in function to the OSI network layer. NPA-NXX North American numbering plan exchange code 12 of 13 8/26/98 2:51 PM

13 PSTN R2 SCP SONET public switched telephone network an analog signaling protocol service control point synchronous optical network SS7 signaling system 7 SSP STP TCAP VoIP WAN service switching point signal transfer point transaction capabilities application part: Transaction capability application part messages are used to support non-circuit related, connectionless information exchange. Among other things, TCAP messages are used to send queries to databases (such as toll-free [freephone] databases), and to return the database response. voice over Internet protocol wide area network [List of Topics] [Tutorial Home Page] Home Page Comment on this tutorial. Home Page Last modified Wednesday, 01-Jul :11:34 CDT Copyright 1998 The International Engineering Consortium 13 of 13 8/26/98 2:51 PM