Voice Over IPv6: Architectures for Next Generation VoIP Networks

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But as you know, you can get a good idea of the concepts faster from the drawings alone then dig into the text for more detail.

What is Voice over IP?

Chapter 1 is a big picture overview that gets into the reasons for both technologies. Chapters 2 and 3 are a tutorial on VoIP.

Chapter 4 delves into the subject of "presence," which along with instant messaging have recently emerged as the new medium of communications over enterprise networks, extranets, and the Internet. It also covers network address translation NAT , which is the process of mapping IP addresses from one group to another. The basics of the IPv6 protocol is covered in Chapter 6. Starting with version 4 IPv4 , which is still the predominant Internet protocol, it shows how version 6 solves some critical problems now and in the future especially with the vastly increased address space.

Chapter 7 was unexpected. Again, lots of detail but good illustrations. The book comes to an end with Chapter 8 that details some issues related to the transition from IPv4 to IPv6. This book sites lots of references so you can get more detail if you need it. I have two main complaints about the book. Minoli does assume you already know quite a bit about this subject. He lost me in a few places when he could have easily given a tidbit of background prerequisites to help out. The good news about authors like Minoli is that they really do know their subject. But the bad news is that they often know it so well they expect you to know all the background they do as well.

Mostly he does a good job with this.

IPv6 Technology

It has an architecture designed for scalability and flexibility and high performances. SER history spans back to the previous century. SER's particular strength is its performance SER runs well even under heavy load caused by large subscriber populations or abnormal operational conditions , flexibility SER's genuine configuration language and module interface allow high degree of customization and interoperability tested and operated against tens of SIP products over the years, including but not limited to Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, xten, and many others.

Yate is a next-generation telephony engine; while currently focused on Voice over Internet Protocol VoIP , its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. Erlang was developed by Ericsson to program ordinary telephone switches, with the goal of making a programming system fault-tolerant and robust.

Particular attention should be given to ensuring that emergency calls on VoIP lines are handled correctly. In some cases, it may not be possible to provide useful or reliable emergency service via VoIP services. In this case, it may be possible to offload emergency calls to a landline service. If reliable emergency service is not available, the handset or VoIP device should clearly be marked with a warning. Some VoIP providers allow emergency calls to be routed via their service; however this is only of use if you are within an area covered by the receiving emergency call centre.

Additionally, a lack of bandwidth or Internet service or a power loss can lead to loss of emergency service via telephone. If a functioning landline handset is nearby, it is advisable to attach a notice to the VoIP device directing emergency calls to the landline. It is much easier for emergency services to assist if they have information on your location which is generally not available via VoIP calls.

Where calls are handled by a gateway administered by yourself or your organisation, and the ability to route certain calls via landline exists, the dialplan for your VoIP gateway should immediately route emergency calls via the landline, and where possible, any active conversations should be dropped if there are not sufficient lines to handle the call.

It is important to find out whether your VoIP provider s offer emergency numbers, what areas they are able to cover and whether they are able to provide enhanced emergency services such as e initiatives which allow providers to transmit information on subscribers to call centres. IVR applications are automated response units which receive input from the user traditionally in the form of DTMF and allow the user to progress along a pre-set interactive path.

For example, a menuing system where the IVR may announce Press 1 for Sales, 2 for Marketting and then await user input. The following section is quite technical, as it deals with complicated infrastructure and the integration of VoIP into core internet technologies. These records are stored in a hierarchical tree beneath the e Records within the e Much like the in-addr.

I'd like to be notified of new arrivals in the following categories.

The agency is then responsible for further smaller delegations where necessary. As a result, it is unlikely that individuals or small businesses will have access to provide ENUM records for their telephone numbers. It also stands to reason that telephone companies would be hesitant to allow telephone calls destined to their customers to instead terminate over the Internet, as much of a phone company's revenue would be derived from call termination fees, charged to carriers for calls terminating on a company's network.

This is unfortunate for end users. For this reason, several free ENUM databases have been created to allow users to share alternate IP-based paths to their phone numbers. Most ENUM aware gateways will allow additional 'search domains' to be configured, providing access to these unofficial databases.

If a malicious or unauthenticated entry is added, an attacker can direct calls for a phone number to an arbitrary VoIP gateway. Consider the example of rival businesses. If Business A were tech-savvy and entered their competitor's number into a public directory, directing calls to their own VoIP gateway, any calls placed via a VoIP gateway configured to query that directory would be directed to Business A's phone, rather than the intended recipient, Business B.

Voice Over IPv6

Some of these directories require voice verification of ownership when you register phone numbers. Others do not require verification or have dispute resolution policies which deal with disputes on a case-by-case basis. Another less serious but still important consideration is the possibility that your information will be harvested from the database, or sold by the database owner.

ENUM information may be valuable to advertisers or phone spammers, as it would provide them with a list of valid phone numbers, and allow them to call for free via IP. Many ENUM trial registries are becoming available. Most of these registries are regional, and often are limited by telephony provider, limited number of trial participants, or a trial fee. Currently has 46 members. From Wikibooks, open books for an open world. The network event component collects call events recorded at the soft switch sites. Call event records can be used, for example, for fraud detection and prevention, and billing.

The provisioning event component receives provisioning requests from upstream operational support services OSS systems such as, for example, for order-entry, customer service and customer profile changes. The provisioning component distributes provisioning data to appropriate network elements and maintains data synchronization, consistency, and integrity across multiple soft switch sites.

The network management component includes a network operations center NOC for centralized network management. Each network element NE e. The NOC uses the events generated by each network element to determine the health of the network and to perform other network management functions.

First, a soft switch site receives an incoming call signaling message from the signaling network. The soft switch site determines the type of call by performing initial digit analysis on the dialed number.


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The soft switch site then queries a customer profile database to retrieve the originating trigger plan associated with the calling customer. The query can be made using, for example, the calling party number provided in the signaling message from the signaling network. This look-up in the customer profile database returns subscription information. For example, the customer profile may indicate that the calling party has subscribed to an account code verification feature that requires entry of an account code before completion of the call. In this case, the soft switch site will instruct the gateway site to collect the account code digits entered by the calling party.

Assuming that the gateway site collects the correct number of digits, the soft switch site can use the customer profile to determine how to process the received digits. For account code verification, the soft switch site verifies the validity of the received digits. In this example, the soft switch site can verify that the account code is valid, but that it requires that an intrastate COSR should be enforced. This means that the call is required to be an intrastate call to be valid. The class of service restriction logic can be performed within the soft switch site using, for example, pre-loaded local access and transport areas LATAs and state tables.

The soft switch would then allow the call to proceed if the class of service requested matches the authorized class of service. The soft switch site then completes customer service processing and prepares to terminate the call. At this point, the soft switch site has finished executing all customer service logic and has a digit dialed number that must be terminated.

To accomplish the termination, the soft switch site determines the terminating gateway. The dialed number i. This termination may be selected based on various performance, availability or cost criteria. The soft switch site then communicates with a second soft switch site associated with the called party to request that the second soft switch site allocate a terminating circuit or trunk group in a gateway site associated with the called party.

One of the two soft switch sites can then indicate to the other the connections that the second soft switch site must make to connect the call. The two soft switch sites then instruct the two gateway sites to make the appropriate connections to set up the call. The soft switch sites send messages to the gateway sites through the data network using, for example, IPDC protocol commands.

Alternately, a single soft switch can set up both the origination and termination. The present invention provides a number of important features and advantages.

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First, the invention uses application logic to identify and direct incoming data calls straight to a terminating device. This permits data calls to completely bypass the egress end office switch of a LEC. This decrease in cost results partially from bypass of the egress. A further advantage for ISPs is that they are provided data in the digital form used by data networks e. Consequently, the ISPs need not perform costly modem conversion processes that would otherwise be necessary. The elimination of many telecommunications processes frees up the functions that ISPs, themselves, would have to perform to provide Internet access.

Another advantage of the present invention is that voice traffic can be transmitted transparently over a packet-switched data network to a destination on the PSTN. Yet another advantage of the invention is that a very large number of modem calls can be passed over a single channel of the data network, including calls carrying media such as voice, bursty data, fax, audio, video, or any other data formats.

Further features and advantages of the invention, as well as the structure and operation of various embodiments of the invention, are described in detail below with reference to the accompanying figures.