VOIP Advantages & Disadvantages
In general, phone service via VoIP costs less than equivalent service from traditional sources but similar to alternative PSTN service providers. Some cost savings are due to using a single network to carry voice and data, especially where users have existing under-utilized network capacity they can use for VoIP at no additional cost. One must note that the maximum upstream in your Internet connection is the final throttle and service is not as good as standard telco services.
VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user. Free VoIP to PSTN services are rare. A notable provider is VoIP User. There are two types of PSTN to VoIP services: DID and access numbers. DID will connect the caller directly to the VoIP user while access numbers requires the caller to input the extension number of the VoIP user. Access numbers are usually charged as a local call to the caller and free to the VoIP user while DID usually has a monthly fee. There are also DID that are free to the VoIP user but is chargeable to the caller.
VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:
Incoming phone calls can be automatically routed to your VoIP phone, irrespective of where you are connected to the network. Take your VoIP phone with you on a trip, and anywhere you connect it to the Internet, you can receive your incoming calls.
Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection.
VoIP phones can integrate with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books and passing information about whether others (e.g. friends or colleagues) are available online to interested parties.
VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than legacy digital PBX ports.
Because IP does not provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, VoIP implementations may face problems dealing with latency (especially if satellite circuits are involved), and jitter. They are faced with the problem of restructuring streams of received IP packets, which can come in any order and have packets delayed or missing, to ensure that the ensuing audio stream maintains a proper time consistency. This problem has been addressed by Ubicom with their StreamEngine Technology.
Another main challenge is routing VoIP traffic to traverse certain firewalls and NAT. Intermediary devices called Session Border Controllers (SBC) are often used to achieve this, though some proprietary systems such as Skype traverse firewall and NAT without a SBC by using users' computers as super node servers to route other people's calls.
Keeping packet latency acceptable can also be a problem, due to network routing time (buffering, switching) and transmission distances (more relevant under satellite links).
Conventional telephones 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. In order to use VoIP during a power outage, an uninterruptible power supply or a generator must be installed on the premises. It should be noted that many early adopters of VoIP are also users of other phone equipment such as PBX and cordless phone bases that also rely on power not provided by the telephone company.
Some broadband connections may have less than desirable reliability. 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 interworking between end points.