Voice over IP also called IP Telephony is a new application service to communicate voice traffic over the broadband network. It is one of the killer applications suitable to implement in broadband NGN. However, there are several issues to be addressed while the telecom operators plan to swift the traditional circuit-switched voice traffic to the IP based broadband NGN. The most critical ones include inter-connection, interoperability and universal services such as number portability and E911 emergency call service. In order to provide good VoIP solution to the incumbent or new telecom operators to implement a reliable NGN, this research presents the inter-exchange center of converged network architecture (IEXCNA) model to help the operators to provide reliable VoIP service to the customers as well as peered networks. First it is necessary to introduce what the system functionality of IEXCNA for Voice over IP service is.
4.1 System Functionality of IEXCNA for VoIP Service
VoIP is also called IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network. Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network - see attached image - to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that's borne by the VoIP user.
4.1.1 Reliability
Conventional phones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages dictating the use of an uninterruptible power supply or generator to ensure availability during power outages. Early adopters of VoIP may also be users of other phone equipment, such as PBX and cordless phone bases, that rely on power not provided by the telephone company. Even with local power still available, the broadband carrier itself may experience outages as well. While the PSTN has been matured over decades and is typically extremely reliable, most broadband networks are less than 10 years old, and even the best are still subject to intermittent outages. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T1 connection.
4.1.2 Quality of Service
Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there is long distances and/or inter-working between end points.
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Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on (TS 185 001, 2005).
It has been suggested to rely on the packetized nature of media in VOIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, the temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer.
RFC3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. RFC3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
4.1.3 Emergency calls
The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department (TS 182 009, 2006).
In the US, at least one major police department has strongly objected to this practice as potentially endangering the public. Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way.
Following the lead of mobile phone operators, several VoIP carriers are already implementing a technical work-around. For instance, one large VoIP carrier requires the registration of the physical address where the VoIP line will be used. When you dial the emergency number for your country, they will route it to the appropriate local system.
They also maintain their own emergency call center that will take non-routable emergency calls (made, for example, from a software based service that is not tied to any particular physical location) and then will manually route your call once learning your physical location.
The United States government had set a deadline, requiring VoIP carriers to implement E911; however, the deadline is being appealed by several of the leading VoIP
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companies.
4.1.4 Integration into global telephone number system
While the traditional Plain Old Telephone System (POTS) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks.
Some allocate an E.164 number which can be used for VoIP as well as incoming/external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider specific short codes.
4.1.5 Single point of calling
With hardware VoIP solutions it is possible to connect the VoIP router into the existing central phone box in the house and have VoIP at every phone already connected.
Software based VoIP services require the use of a computer, so they are limited to single point of calling, though handsets are now available, allowing them to be used without a PC. Some services provide the ability to connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any location with an open hotspot.
4.1.6 Mobile phones & Handheld Devices
Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones) or VoIP is implemented over legacy 3G networks.
However, "dual mode" handsets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.The first mobile VoIP solutions were Fring and Truphone. Phones like the Nokia E60, E61 have been the first "dual mode" handsets capable of delivering mobile VoIP.
With more and more mobile phones and handheld devices using VOIP, the nicknames of
"MoIP" and MVoip (Mobile VoIP)have been attributed to these mobile applications.
Handheld Devices are another type of medium whereby you can use VoIP services. Since most of these devices are limited to using GSM/GPRS type of communication mediums, almost all of the handheld devices use WiFi of some sort. Another addition to handheld devices are ruggedized barcode type devices that are used in warehouses and retail environments. These type of devices rely on "inside the 4 walls" type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.
4.1.7 Security
The majority of consumer VoIP solutions do not support encryption yet. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open
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source applications; however such security through obscurity has not proven effective in the long run in other fields.
Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a consumer level. The existing secure standard SRTP and the new ZRTP protocol is available on Analog Telephone Adapters(ATAs) as well as various softphones.
It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream (TS 181 010, 2005).
4.1.8 Caller ID
Caller ID support among VoIP providers varies, although the majority of VoIP providers now offer full Caller ID with name on outgoing calls. When calling a traditional PSTN number from some VoIP providers, Caller ID is not supported.
In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, making it appear as though they are calling from a different number. Business grade VoIP equipment and software often makes it easy to modify caller ID information. Although this can provide many businesses great flexibility, it is also open to abuse.
4.1.9 Legal issues of VoIP Service
As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, the government is becoming more interested in regulating VoIP in a manner similar to legacy PSTN services, especially with the encouragement of the telephone monopolies/oligopolies in a given country, who see this as a way to stifle the new competition.
In the U.S., the Federal Communications Commission now requires all VoIP operators who do not support Enhanced 911 to attach a sticker warning that traditional 911 services aren't available. The FCC recently required VoIP operators to support CALEA wiretap functionality and Lawful Interception. The Telecommunications Act of 2005 proposes adding more traditional PSTN regulations, such as local number portability and universal service fees. Other future legal issues are likely to include laws against wiretapping and network neutrality (TS 187 005, 2006).
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4.2 Call Flow of VoIP Service in IEXCNA Model
The existing network operators to run VoIP service has encountered the following problems such as:
– Service traffic Inter-connection across different network operators – Provide Emergency call service such as E911 to the end users – Provide Number Portability service to the end users
To solve these problems and meet the customer’s requirements, Inter-Exchange Center of Converged Network Architecture (IEXCNA) Model is able to provide solution to achieve two objectives. First, IEXCNA is able to provide network inter-connection and interoperability of VoIP Services for the Type 2 (E.164/non-E.164) VoIP operators.
Second, it can provide a complete suite of VoIP services including Lawful Interception (LI), E119 emergency call service, Number portability (NP) Query, etc.
4.2.1 System Architecture of IEXCNA model
The IEXCNA model is based on IMS Core to implement the solution. The system architecture of IEXCNA is shown in Fig. 4.2-1.
Figure 4.2-1 System Architecture of IEXCNA
The IEXCNA model contains a Services Delivery Framework. For the deployment of
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VoIP service, the Services Delivery Framework provides interconnection function for connecting type 2 ISP and type1 PSTN operators. It also contains a NP/ENUM database to provide NP Querying service for all incoming and outgoing calls. The ENUM is dedicated for the Private ENUM International Peering (Faltstrom and Mealling, 2004).
The E119/E110 Services is implemented in the application servers. The SIP server provides the following functions:
– REGISTER allows either the user or a third party to register contact information with a SIP server.
– INVITE initiates the call signaling sequence.
– ACK and CANCEL support session setup.
– BYE terminates a session.
– OPTIONS queries a server about its capabilities.
Some of the important SIP functional entities are listed below.
User agent performs the functions of both a user agent client, which initiates a SIP request, and a user agent server, which contacts the user when a SIP request is received and returns a response on behalf of the user.
acts as both a SIP client and a SIP server in making SIP requests on behalf of other SIP SIP proxy clients. A SIP proxy server may be either stateful or stateless. A proxy server must be stateful to support TCP, or to support a variety of services. However, a stateless proxy server scales better (supports higher call volumes).
Registrar is a SIP server that receives, authenticates and accepts REGISTER requests from SIP clients. It may be collocated with a SIP proxy server.
Location server stores user information in a database and helps determine where (to what IP address) to send a request. It may also be collocated with a SIP proxy server
Redirect server is stateless. It responds to a SIP request with an address where the request originator can contact the desired entity directly. It does not accept calls or initiate its own requests.
IEXCNA Carriers Interfaces include to Type 1 PSTN/PLMN SS7 E1 link and to Type 2 ISPs SIP interface of E1 Link and 100/1000 Ethernet Interfaces. Internal interfaces provide functions to Support LI (Lawful Interception), to Support EMG 119, to Support NP Query, to Support ENUM Peering and Billing Clearinghouse Function for Carrier Level Settlement of Call Data Records Audit & Dispute Resolution.
When a call is picked up by IEXCNA, it performed the following call handling functions. Step1. IEXC performs Number portability Query function to identify the recipient network. Step2. Then B-Numbering analysis is performed by IEXCNA to find out the routing to destination networks. Step3. If it is an emergency call, then it will be routed to emergency call service center for further processed. For a normal call, the call is put on the transmission line according to routing path analyzed from Step 1 and 2.
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Figure 4.2-2 IEXCNA model of Network Inter-connections for VoIP
The IMS of IEXCNA model provide SIP based call routing and service delivery for the user calls originated or terminated in the VoIP operator’s network where it is connected to VoIP IEXCNA. IEXCNA provide inter-connecting point among ISP and PSTN for traffic exchange and call routing. The number portability (NP) database is to analyze the incoming called party number and mapping with network routing number (RN) of the recipient network prefixed in the originated called party number. As such, the incoming call can be correctly routed to the destination. The numbering plan and routing table information are generated in the IEXCNA NP database by storing all national numbering plan and corresponding network codes. Therefore, no matter what network the called party number belongs to, the IMS of IEXCNA can query the NP database to find out the correct network destination and call routing information. The routing query is described in the Fig. 4.2-3.
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Figure 4.2-3 The National Numbering Plan and Routing Table Database
4.2.2 Call Flow Analysis of the IEXCNA model network
With the routing table and numbering plan database, the calls can be handled correctly and efficiently to provide efficient service. The routing flow of call from Type 2-A operator to Type 2-B operator is shown in the Fig. 4.2-4.
Figure 4.2-4 Type 2-A to Type 2-B Call via IEXCNA
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The routing flow of call from Type 2-A operator to Type I-PSTN operator is shown in the Fig. 4.2-5.
Figure 4.2-5 Type 2 A to Type 1 PSTN Call via IEXCNA
The routing flow of call from Type I-PSTN operator to Type II-A operator is shown in the Fig. 4.2-6.
Figure 4.2-6 Type 1 PSTN to Type 2 A Call via IEXCNA
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4.2.3 Signaling Flow Analysis of the IEXCNA model network
With the IP address for signaling routing, the signaling messages flow of SIP based can be handled correctly and efficiently to provide efficient service. The routing flow of signaling from Type 2-A operator to Type 2-B operator is shown in the Fig. 4.2-7.
SIP Server A
Figure 4.2-7 Signaling Message Flow Type 2 A to Type 2 B via IEXCNA
The signaling message flow of call from Type 2-A operator to Type 1-PSTN/PLMN operators is shown in the Fig. 4.2-8.
Figure 4.2-8 SIP Signaling Message Flow Type 2 A to Type 1 PSTN/PLMN via IEXCNA
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The signaling message flow of call from Type 1-PSTN/PLMN operator to Type 2- operators is shown in the Fig. 4.2-9.
SIP Server A
Application Server A
NP Query ENUM
CDR Server 2. Invite
4. Invite
7. Invite 8.. 100 Try 3. 100 Try
CDR Info 5. 302 Moved Temp
6. Ack
Proxy Exchange
Service 1. IAM
Trunk/Media Gateway
Session Border Controller (SBC)
9. Invite Type 2 Carrier(s)
[SIP]
Type 1 Carrier(s)
[SS7]
media
Figure 4.2-9 SIP Signaling Message flow Type 1-PSTN/PLMN to Type 2 A via IEXCNA
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4.2.4 OSI 7-Layer Model: Internet & VoIP/SIP Peering
The Inter-Exchange Center connection model provides network protocol peering according to OSI 7-layer model. The network and SIP peering of VoIP service is shown in the Fig. 4.2.10.
Figure 4.2-10 Internet & SIP Peering for VoIP service
The network of Inter-Exchange Center connection model can support the following functions.
– PSTN and PLMN Routing Info – Registry Service
– Provisioning transactions – Routing query/response
– SIP signaling message handle, SIP Identity, Signaling Proxy and Validation Registry – Call Data Record (CDR) output and settlement, accounting and clearing
– RTP transportation of multi-media stream – IP packets incoming and outgoing
The detailed functional blocks of IEXCNA model for VoIP can be seen in the Fig. 4.2-11
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Figure 4.2-11 The functional blocks of IEXCNA model for VoIP service
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