Softswitch for Voice Tandem Service: Broadband and Narrowband Interworking

Transcription

1 for Voice Tandem Service: Broadband and Narrowband Interworking James Yu, Ph.D. Sea Light, Inc. Telecommunications and ing Consultation Naperville, IL USA ABSTRACT This paper presents the use of softswitch for voice tandem service that is to transport voice traffic between PSTN over a packet network. The industry is moving in the direction of integrated voice and data services, and the integration is built on a packet core network, ATM or IP. As a result, we see the growing needs to carry voice traffic over a packet network. Voice tandem service has been frequently discussed but it is not widely deployed yet. A major issue is the signaling scheme between the circuit world (SS7) and the packet world (ATM/IP). The question is how to carry SS7 information in the ATM/IP signaling messages, and vice versa. This paper discusses several approaches, along with the pros and cons of each approach. The solution in this paper is targeted for the carrier network where the capacity requirement is at least 1,000,000 busy hour calls. Solutions applicable to the enterprise network are usually not scalable to support the carrier network. Although the ideal solution is Broadband ISUP (B-ISUP), we do not consider it a practical one due to its complexity and a lack of vendor support. A more practical one is the combination of SS7 signaling and Q.931-like (ATM UNI or H.323) signaling. We see the need for industry standards to address this issue so that softswitches from different vendors can talk to each other; otherwise, voice tandem would be limited to a single carrier s network. Other issues, such Quality of Service (QoS), echo cancellation, and voice codec (and compression), are important only if we can resolve the signaling issue first. Page 1 of 10

2 1 Introduction This paper describes the use of softswitch for voice tandem service that is frequently discussed but not widely deployed yet. The concept of softswitch was first introduced at Bell Laboratories, Lucent Technologies in mid 90 s, and it was formally adopted by the industry with the establishment of International Consortium (ISC) in The terminology of this paper follows the North American (ANSI) signaling standards (e.g., SS7), and the concept is applicable to international standards as well (e.g., C7 in ITU-T). The International Consortium provides the following definition [1]: (a.k.a. call agent, call server or media gateway controller) is a device that provides: 1. Intelligence that controls connection services for media gateways, and/or native IP endpoints. 2. The ability to select processes that can be applied to a call. 3. Routing for a call within the network based on signaling and customer database information. 4. The ability to transfer control of the call to another network element. 5. Interfaces to and supports management functions such as provisioning, fault, billing, etc. The standard softswitch architecture is illustrated in Figure 1. From the definition and diagram, softswitch is a logical unit (could be many physical boxes) that controls multiple media gateways and interfaces with five networks (access, PSTN, ATM/IP core, SS7, and OSS). For end-users, the need is to place a call through the network., after receiving a call request, designates an endto-end circuit to the destination, allocates the resource (i.e., bandwidth), and routes the call. The concept of calls expands beyond traditional voice calls; it is a connection request between two end-points which could be phones (voice or IP), PBX (ISDN or IP), IAD (ATM or IP), workstations, servers (database, video, announcement, etc.), or even IP appliance. During the process of calls, softswitch may request additional information to route calls (e.g., 1-800/900) and the query requests are sent to the SS7 network (SCN) or an IP-based network (via LDAP). 2 Application Overview Although softswitch has the potential to replace local, toll, and gateway switches [2], its primary application today is Internet offload as illustrated in Figure 2. SS7 Access IP-based Resource Application Server OSS PSTN Packet (IP or ATM) Figure 1. Architecture Diagram Page 2 of 10

3 SS7 Internet Local Tandem ISDN-PRI RAS (modem pool) There are several factors for the success of this application:? Simplicity. The only feature that softswitch needs is SS7 to ISDN-PRI signaling. Both protocols and their interworking are well defined and can be purchased from 3 rd party software vendors.? Low Reliability Requirement. Because this is a pure data service, the availability/reliability requirement is only 2-9 s (or a few hours of downtime per month). For local voice service with 911 requirements, 5-9 s (less than 5 minutes of downtime per year) is mandatory.? Cost. There are no issues with voice quality, echo cancellation, or quality of service. Therefore, there is no need for expensive DSP cards, which would at least double the cost per DS0 port. The low reliability requirement makes the simplex solution acceptable, which also lowers the cost.? Regulation. The most important economic factor is the reciprocal charge for the traffic from one carrier s tandem switch and another carrier s media gateway. However, this factor no longer exists due to regulation changes. Figure 2. for Internet Offload Service Due to the regulation (such as E911 and CALEA) and reliability requirements, it is unlikely that softswitch could compete with legacy circuit switches in the near future. The advantages of softswitch are in (1) integrated services, (2) service creation platform, and (3) network management. We will have a separate paper to discuss the applications of softswitch for local services. Voice toll and gateway services also require significant up front development effort. The common denominator for all voice services is the voice tandem service. 3 Voice Tandem Service From an end-user perspective, the network provides a connection to the destination identified by a given address. In the voice world, the address is a telephone number following the E.164 format. When the voice is converted to packets, the address maps to an IP address (IPv4 format) or an ATM address (NSAP and/or VPI/VCI). During the process of establishing the connection, the network uses other addresses, such as SS7 Point Code (PC) and Ethernet MAC address. 3.1 Legacy Tandem According to a report from the Association for Local Telecommunications Services (ALTS) [3], there are 49M dial-up lines served by carriers in the United States. With the growing popularity of broadband access (wireless 3G, DSL, and Cable modem), the market for dial-up is saturated, if not declining. needs to explore and identify the next killer application. The telephone network for tandem service is mostly based on SS7 signaling, a and the call setup is illustrated Figure 3. a We acknowledge that MF trunks are still used on the network. However, softswitch only needs to interface with PSTN via SS7. Page 3 of 10

4 STP STP STP Local (A) Toll/Tandem Local (B) When the origination local switch receives a call request with the destination phone number, the switch sends an SS7 message () to Signaling Transfer Point (STP which looks for the Destination Point Code (DPC) in the message and forwards the message to the tandem switch. The tandem switch confirms the availability of the incoming circuit identified by the Circuit Identification Code (CIC), and allocates an outgoing circuit for the call. The tandem switch then composes a new message and puts the ID (CIC) of the outgoing circuit in it. The new message is then sent to the next switch via STP. As the message traverses through the network, an end-to-end circuit is established for the call. Figure 3. Legacy Tandem Architecture 3.2 Virtual Circuit Connection In the telecom history, the introduction of voice switches is a major milestone because it allows network resource to be shared by all users. In a switched network, each circuit is no longer dedicated to a specific user. When the network resource is shared, the utilization is increased (by a factor of 10 or more) and the cost goes down accordingly. However, the Public ed Telephone (PSTN), as the name implies, is designed for voice traffic only. As the industry moves to the packet core network (ATM or IP), network resource needs to be shared by voice and data users. The complete call flow diagram is illustrated in Figure 4. ACM ANM REL RLC Tandem TALKING ACM ANM REL RLC Origination Termination Figure 4. SS7 Call Flow Diagram Our focus is on the establishment of an endto-end circuit, which is the function of. For the rest of this paper, we discuss call flows only for and related broadband signaling messages. Another trend in the telecom industry is to move from a hierarchical network to a flat network. The hierarchical network requires significant effort in network traffic engineering because it is not easy to allocate network resource from one switch to another. To alleviate this problem, we see expensive Digital Cross Connect (DACS) on the network. b The packet network is a flat network and bandwidth can be allocated on demand. It completely eliminates the need of DACS. As a result, it is expected to have significant savings on network deployment (no need for expensive DACS) and network management (less effort on traffic engineering.) fills in the gap to bridge the circuit world (PSTN) and the packet world (IP/ATM) as illustrated in Figure 5. b A high capacity DACS (multiple DS3, OC-3, and OC-12) costs more than $1M. Page 4 of 10

5 SS7 Local ATM/IP Core Local Figure 5. Architecture of for Voice Tandem Service The protocols for the voice tandem network are summarized in Table 1. Table 1. Protocols of Tandem Pont A Point B Protocols 1 PSTN SS7 2 H.248 (Megaco) 3a SS7 3b SIP 4 UNI, PNNI, or H PSTN Bearer Channel (no messages) In the circuit world, a circuit is identified by CIC. In the ATM world, a [virtual] circuit is identified by VPI/VCI. In the IP world, a [virtual] circuit is a UDP port. When the ingress softswitch gets an message, it needs to allocate a virtual circuit to the call. When the egress softswitch gets an message, it needs to confirm the availability of the specified virtual circuit and allocate a TDM circuit to PSTN. The H.248 (Megaco) standard addresses the interoperability issue between softswitch and media gateway. However, there are still two major issues with voice tandem service over a packet network: 1. How do softswitches communicate with each other? The popular scheme is SS7. When the ingress softswitch selects the outgoing circuit (CIC), it sends to the egress softswitch via the SS7 network. The problem with this approach is the increase of traffic on the expensive SS7 link. The ideal solution is to send SS7-like messages over the packet network (ATM/IP) [4]. We are aware of a SIP-to-ISUP-mapping proposal to IETF [5], and the discussion of using this scheme [6][7]. However, due to the complexity of ISUP, we do not see a well-defined mapping scheme between and INVITE, plus many other SS7 messages. Therefore, we are probably years away from seeing an interoperable solution from different vendors. This paper uses the SS7 approach, and the discussion is applicable to the SIP approach. INVITE (SIP Message) IP SS7 (a) SIP (packet) approach (b) SS7 (circuit) approach Figure 6. Communications between es 2. How do SS7 messages interwork with ATM/IP signaling messages? The problem is that the packet (ATM/IP) protocols do not have the ability to carry all SS7 information, and the SS7 protocols do not have the ability to carry Page 5 of 10

6 packet (IP/ATM) information. Section 4 has a detailed discussion on this issue. 3.3 Scalability and Capacity Requirements The scope of this paper is on carrier network, instead of an enterprise network. In the carrier network, a tandem switch is connected to multiple class 5 (local) switches, and each local switch has thousands (or tens of thousands) of circuits. On the economical scale of a typical carrier network, a single softswitch is required to be scalable to process 1,000,000 busy hour calls (BHC), or 278 calls/sec. If a vendor s solution requires multiple softswitches to support 1,000,000 BHC, this solution would to too expensive to deploy (cost per DS0) and too costly to manage. As a general rule for traffic engineering, we assume 3 minutes (180 sec) duration for a voice call. With that, we can calculate the number of non-blocking calls: 1,000,000 / 3,600 * 180 = 50,000 A softswitch controls multiple media gateways, and the cluster of media gateways needs to be scalable to support 50,000 voice circuits (DS0). A single media gateway for carrier networks usually supports 8,000 to 32,000 circuits. If a media gateway supports less than 8,000 circuits (DS0), it is unlikely to be cost-effective for a carrier network. We can also calculate the number of SS7 (V.35) links to support 1M BHC: SS7 messages per call: 5.5 msg/call Per tandem call (in+out): 5.5 * 2 = 11 Average message size: 30 bytes Number of calls per V.35 link (56K): 56,000 / (11 * 30 * 8) = 21 calls/sec Number of SS7 links: [ 278 / 21 ] = 14 In summary, a single softswitch for tandem service on a carrier network needs to support 1M BHC, 50,000 non-blocking calls, and 14 SS7 links. 4 SS7 and Packet (ATM/IP) Interworking There are no standards to address the issue of SS7 and ATM/IP interworking, and each softswitch vendor has its own solution to this problem. As a result, softswitches from different vendors cannot talk to each other. This section presents several signaling schemes and discusses their pros and cons. 4.1 Single Solution The simplest solution is to eliminate the need for softswitches to communicate with each other. In this case, each softswitch on the network can control every media gateway and all softswitches on the network share a centralized database that contains information of all circuits on media gateways Circuit Setup Request Softswit ch (ingress) 3. Circuit Circuit 4. Circuit Setup Request Figure 7. Call Flow Diagram of Single Solution The advantage of this solution is simplicity and the disadvantage is not scalable. For a softswitch with 1M BHC capacity, this architecture could support a network of 50,000 circuits (DS0) only. This solution is acceptable for small carriers offering data dial-up services c on a regional network. 4.2 PVC Solution In the case of PVC solution, all virtual circuits are predefined and identified by VPI/VCI. There is a one-to-one mapping of each PVC and CIC, and there is no need for c For data dial-up services, the average call duration is 20 minutes, and it increases the number of concurrent calls by 700%. (egress) Page 6 of 10

7 media gateways to exchange signaling information. interfaces for two boxes, it is unlikely we could see standards for all three boxes in SS Circuit Setup Request (ingress) 3. Circuit ATM Core 5. Circuit Setup Request (egress) 6. Circuit Figure 8. Call Flow Diagram of ATM PVC As illustrated in Figure 8, the PVC solution is identical to the circuit solution described in Section 3.1. does not see any difference between calls from PSTN or from ATM/IP. The media gateway functions as a virtual Digital Cross Connect (DACS) between physical trunks (TDM circuit) and virtual trunks (ATM/IP virtual circuits). d The major advantage of the PVC solution is avoiding the interworking issue between broadband (ATM/IP) and narrowband (SS7) signaling, and it makes development work relatively easy. The major disadvantage is management and utilization of the ATM/IP trunks. Because PVCs are dedicated to the voice service, they cannot be used for data service even when the voice traffic is low. The nature of PVC defeats the purpose of integrated voice and data network. When the number of PVCs grow to thousands (or hundreds of thousands), it will be difficult to provision and manage the [virtual] circuits. A powerful management tool could resolve this network management problem where the management of PVC is similar to the management of TDM circuits in today s PSTN. The tool has the capability to provision and manage (OA&M) thousands (or hundreds of thousands) of circuits by a friendly Graphical User Interface (GUI). Note that the PVC provision also includes the ATM core network. In other words, the tool needs to manage softswitch, media gateways, and ATM core switches. As the industry has problems standardizing d There is no formal definition of trunks. In this paper, a trunk is simply a group of circuits (DS0). Many people refer to trunk as T1 or E1. the near future. Unless a vendor could dominate both softswitch and ATM markets, the PVC solution is considered unacceptable due to the issues of interoperability, scalability, and network management. 4.3 SVC (B-ISUP) Solution The standard of Broadband ISUP (B-ISUP) was established several years ago, and it was designed to provide the broadband (ATM) and narrowband (SS7) interworking. The call flow diagram is illustrated in Figure Circuit Setup Request (ingress) 4. Call Setup Request 3. (B-ISUP) ATM Core (egress) 7. Figure 9. Call Flow of B-ISUP 5. Circuit Setup Request 6. Circuit From the standard perspective, B-ISUP is the ideal solution for the softswitch tandem application. However, there are no ATM switch or media gateway vendors supporting B-ISUP. A possible reason is its complexity Page 7 of 10

8 (see Figure 10). It is likely that this solution would exist on paper only. B-ISUP MTP3b SSCF (Q.2140) SSCOP (Q.2110) ATM CS AAL5 ATM Figure 10. B-ISUP Protocol Stacks 4.4 SVC (ATM UNI) Solution Instead of B-ISUP (or NNI), almost all ATM switch and media gateway vendors support UNI (3.0/3.1/4.0). The call flow diagram for SS7 and ATM UNI is illustrated in Figure 11. This solution solves the issue of bandwidth on demand, and provides a full integrated voice and data services on the packet core network. However, it has a major issue with the cross-office delay which is the time from the ingress softswitch getting the message to the egress softswitch sending out the message (from Step 1 to Step 9). The requirement for cross-office delay is usually set at 300ms, where 150ms is allocated to the network and 150ms to both softswitches. The number of messages involved in the call flow, especially for those across the packet network, make it difficult, if not impossible, to meet the requirement. 4.5 Backward SVC Setup To solve the issue of cross office delay, we propose a solution of backward SVC setup as illustrated in Figure 12. SS7 SS Circuit Setup Request 9. Circuit 4. Circuit Setup Request 5. Circuit 2. Circuit Setup Request 5. Circuit 7. Circuit Setup Request 8. Circuit (ingress) (egress) (ingress) (egress) 7. SETUP 8a. ALERT 8b. CONNECT 3. SETUP 4a. ALERT 4b. CONNECT ATM Core ATM Core Figure 12. Call Flow Diagram of Backward SVC Setup Figure 11. Call Flow Diagram of ISUP and ATM UNI As discussed earlier, the ATM/IP core network supports both voice and data traffic, and voice traffic has priority over the data traffic. As a result, voice requests will always go through the network. With this Page 8 of 10

9 assumption, the egress softswitch can send out without waiting for the confirmation of SVC setup. Step 5 (Circuit Request ) is a request for a TDM circuit to PSTN, and not for a virtual circuit from ATM/IP. When the egress switch receives the ACM message, it will then cross connect the virtual circuit with the PSTN circuit. The ingress switch also uses the ACM message to trigger the circuit cross connection. 4.6 IP/H.323 Solution The call flow diagram of H.323 over IP is the same as ATM UNI, as they both follow the Q.931 signaling standard. However, IP has an additional challenge for performance and scalability. Each voice circuit is identified by a UDP port, which is tied to a socket. For each call setup, the media gateway needs to open a (or multiple) socket. After the call completion, the media gateway needs to close the socket. problem, and softswitches from different vendors cannot talk to each other. From a theoretical perspective, we like the B-ISUP solution. However, we understand the complexity of its implement and a lack to support by the vendors. Our recommendation is the reverse ATM SVC (or H.323) setup scheme as described in Section 4.5. We also identify a performance issue for VoIP due to the UDP socket implementation. Our conclusion is that we cannot use UDP port as the circuit ID because it will require a one-to-one mapping between sockets and calls. Unless the industry could solve the socket issue, VToA will prevail over VoIP on the carrier core network. We did an experiment on a 300MHz Solaris machine in The result shows that we can exchange 50,000 msg/sec with one socket. If we open a socket for each message, the performance goes down to 500 msg/sec. If the media gateway needs to open one UDP socket for each call, we will not have a scalable solution. As discussed earlier, a media gateway needs to support 8,000 or more concurrent calls. We are aware of some proprietary implementations that use one socket for all calls. This is a significant improvement in performance but it will have an interoperability issue. 5 Conclusion The industry is moving in the direction of integrated voice and data services, and carriers are converging the voice and data traffic on the packet network (ATM or IP). Therefore, voice tandem service, transporting voice traffic over a packet network, is a must for this converged network. This paper presents several different schemes to address the interworking issue between the packet network and the circuit network. Our conclusion is that the industry does not have a sound and practical solution to this Page 9 of 10

10 References [1] International Consortium, paper_4.pdf [2] E. Krapf, Will Next-Gen es Conquer the PSTN? Business Communications Review, February 2001, pp [3] The State of Local Competition, The Association for Local Telecommunications Services, February [4] Framework Architecture for Signaling Transport, IETF RFC 2719, October [5] ISUP to SIP mapping, <draft-ietf-sippingisup-00.txt>, November 2001 [6] A. R. Modarressi and S. Mohan, Control and Management in Next-Generation s: Challenges and Opportunities, IEEE Communications, October 2000, pp [7] M. Mampaey, TINA for Services and Advanced Signaling and Control in Next- Generation s, IEEE Communications, October 2000, pp Biography James Yu is currently an independent consultant for service and product development of telecommunications and information technology. Between February 2000 and October 2001, he was the director of Technology at ARBROS Communications, responsible for the network architecture/design, equipment evaluation and selection, service development, product testing, and field support of integrated voice and data services to ISP and business customers. He worked at Bell Laboratories (AT&T and Lucent Technologies) for 15 years, working at the development organizations of 5ESS, ATM switch, ATM-based toll switch, and softswitch. Dr. Yu received his B.S. in Electrical Engineering from National Taiwan University, M.S. and Ph.D. in Computer Sciences from Purdue University, West Lafayette, Indiana. Page 10 of 10