Voice over Internet Protocol
Introduction
What is VoIP? VoIP stands for Voice over Internet Protocol. Wikipedia states that it is also referred to as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. Like its many names, VoIP serves many functions. It lets users make and receive phone call over the Internet, transporting voice traffic alongside data traffic such as instant messages (IMs) and e-mail.
This technology breaks down voice communication into tiny “packets.” These packets are sent over the Internet alongside with data. (2005) explains that when this process occurs, the voice packets are treated like any other piece of data-e-mail, Web pages or the text from an IM chat. Once that packets have reached their destination, they are separated from the data and reassembled to form a real-time streaming audio file. VoIP calls share space on the network with everything else and are routed according to an IP address.
The issues. According to (2004), VoIP is troublesome for several reasons. He found the following reasons
The first one would be that it violates the first law of intelligent computing: let someone else do your computing. The telephone company does phones, so why would you want to? (2004) stresses that WANs are prone to outages. You can’t possibly pick up the phone and always expect a dead tone. Even state that Power Outages is one of the drawbacks of VoIP.
The second problem with VoIP is that it eats up network bandwidth. If you plan to install VoIP, you will have to double or triple the bandwidth on every leg of your network. This is not a small expense.
The third problem is that you have to carefully program your network switches to allocate enough bandwidth to VoIP so that heavy Internet traffic does not crash your phone service. This brings about another problem: finding technicians and network engineers who understand the intricacies of such switch programming and monitoring.
But perhaps the biggest problem with VoIP is that it violates another commonsense law: life and safety issues come first. Fire alarms and telephones come first and should be handled by a separate system not in any way dependent on the district WAN. With VoIP, if the network goes down, so do the phones and the e-mail — the two primary means of communication today.
QoS for VoIP
According to , in a 2002 journal article, nearly all enterprise network managers have avoided serious involvement with quality of service (QOS), because they have not implemented voice-over-IP (VOIP) systems. As long as they could keep premises voice traffic off the LAN and separated from bursty data flows, they could skip the packet classification, queue management and other tedious QOS details. Where voice and data traffic flows have been merged and need to be controlled, usually where they share access to a frame relay or IP WAN, QOS appliances have made the task as simple and effective as possible.
further explained that network managers will not be able to avoid QOS issues much longer. For one thing, the current generation of TDM PBXs appears to be the last; as these expire, IP telephony systems will replace them. QOS is required to keep VOIP traffic running smoothly over the LAN infrastructure alongside existing data traffic –at least the Layer 2 and 3 packet marking and queue management mechanisms–even in the most over-provisioned data networks.
Replacing hubs with switches, classifying packets and managing switch queues are the three basic changes required for successfully running VOIP traffic on the LAN. The packet classification and queue management processes controls the traffic passing through VOIP because hubs cannot reliably support VOIP.
Most IP telephones will classify their own traffic, by marking the class of service (COS) bits. These are part of the 802.1 Q/p portion of the Ethernet (Layer 2) packet header. The COS bits tell the switch how to queue the packets when these marked packets hit the first LAN switch.
As packets flow from the LAN edge into the data network core, the Layer 2 COS markings are mapped to Layer 3 TOS and DiffServ, and additional queues and queue mechanisms are often added. Even though these portions of the data network typically include l00-Mbps or 1-Gbps links and more powerful Layer 3 switches and routers, the possibility of VOIP being disrupted by momentary congestion increases as more traffic is aggregated.
TOS and DiffServ must be taken into account when converting from Layer 2 to Layer 3 packet classification, because older LAN switches that can handle TOS may not be able to deal with the additional DiffServ bits. When traffic must transit a device that can’t read TOS or DiffServ, then access control lists (ACLs) are used to steer the traffic through them based on port numbers, IP addresses or other factors. According to Russell Sellers (2002), senior customer engineer with remote network management service provider NetSolve, “You use the ACL mechanism to say: Only allow these packets to go to this destination, or only allow packets from this destination to that destination.”
SIP
The two major competing standards for VOIP are the IETF (Internet Engineering Task Force) standard SIP (Session Initiation Protocol) and the ITU (International Telecommunication Union) standard H.323. SIP is actually more widely adopted but in backbone voice networks, H.323 is the standard of choice. SIP is a useful tool for the “local loop” and H.323 is like the “fiber backbone”.
(2003) states the following benefits expected that are derived from using SIP:
Convergence cum cost-cutting: Because SIP is based on IP, vendors predict that enterprises will find it easier to combine their voice and data systems into a single IP infrastructure. This will improve ROI by requiring fewer people to operate and streamlining move and change procedures with Web-based self-provisioning and support management–whether enterprises opt for premises-based or hosted IP solutions.
Personal productivity: SIP’s openness and flexibility are expected to give end users more features, as well as more specific control over them. The most talked-about examples are presence and instant messaging (IM), although neither of these technically require SIP protocols, and non-SIP versions of each are already in the market. Nonetheless, many vendors are building or planning to build support for these and other features with SIP.
Future functions: SIP is designed to let much more information be “signaled” than basic call set-up and tear-down messages. The possibilities have some developers already imagining new uses for SIP, especially in machine-to-machine communication applications. Such process automations are could gradually alter the way we work and live, much as fax and then email have become part of our lives.
How does SIP get through a firewall or NAT?
SIPKnowledge states the following: “There are several possible approaches to SIP-capable firewalls. One of the difficulties is that, unlike for, say, HTTP, connections are originated both by hosts inside and outside the firewall. A likely arrangement is that a SIP proxy sits “on” the firewall and relays SIP requests between the Internet and the intranet. This proxy would also open up the necessary ports in the firewall to let audio and video flow through, for example using Socks V5. (Such server would normally be referred to as ALG (App. Layer Gateway)). As an alternative, if a firewall or NAT allows outgoing TCP connections, the inside client can open up a TCP connection to an outside proxy. All outgoing and incoming calls would then be handled by that TCP connection. (The client would still have to use SOCKS or similar mechanism to convince the firewall to let RTP packets through.)”.
Vonage and SIPGate
According to , Vonage is “a commercial voice over IP (VOIP) network and SIP company that provides telephone service via a broadband connection. In order to use the service, customers must purchase or use a branded “VOIP router” or a phone adapter that connects to their main router or broadband modem. In addition, an upload speed of 30-90 kbit/s as well as reliable/QoS optimized connection is necessary to make calls without substantial lag or jitter. Subscribers are permitted to choose any number in the country of the service they subscribe to for their primary line. Whether it be Vonage US, Canada or UK, subscribers may choose from any area code regardless of their actual residence. Subscribers also have the option of obtaining additional “virtual numbers” for monthly fee.”
However, Vonage doesn’t offer phone numbers in every area code in the US despite it supports porting a telephone number via the FCC’s (Federal Communications Commission) Local Number Portability.
Residents from US, Canada and UK are the only subscribers of Vonage. In relation to this, subscribers may still plug into the internet anywhere in the globe using routers with phone ports. Wikipedia also mentioned that Vonage also offers “a USB phone adapter that connects a telephone to the USB port of a computer that has internet service, giving it a dial tone and a normal interface to the worldwide telephone network”.
However, it has been reported that Vonage lines has difficulty operating Fax machines. This may either be a regular Vonage line or dedicated Fax line. There has also been reports of difficulties with Tivo and residential alarm systems.
SIPGate on the other hand is based in Germany. states that SIPGate also operates in Austria and UK and “their service offers free calls to other IP phones, chargeable calls to landlines and mobiles and free geographical and non-geographical incoming phone numbers. Users may either use a piece of SIP client software on their computer or a hardware IP phone with the service. IP phones and other associated hardware can be purchased through SIPGate’s online store, which sells hardware by manufacturers including Grandstream and UTStarcom. In January 2006, they began providing their customers with the ability to send SMS text messages through their website.”
Conclusion
(2006) concludes that “VoIP technology enables a much broader range of technical and business approaches than were feasible in the PSTN world. On the one hand, this new technical reality heralds a new era of innovation and flexibility for users, while on the other hand it makes difficult–or, more likely, impossible–the task of mapping traditional social policy goals and constructs from the PSTN world to the VoIP world in a straightforward manner. Many current social policy goals should be preserved in one form or another and new policy goals should be considered over time. With these thoughts in mind, it seems we would be well-served by a more flexible, rapid, and innovative method of mapping such goals onto the increasingly heterogeneous world of telecommunications.”
References
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