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 in the absence of a uninterruptible power supply or generator. 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 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 T-1 connection.
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 are long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.
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, 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.
Difficulty with sending faxes
The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed for fax transmission. (They are designed to digitize an analog representation of a human voice efficiently, but the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax “sounds” simply don’t fit in the VoIP channel.) An effort is underway to remedy this by defining an alternate IP-based solution for delivering fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as a message switching system, which does not need a real-time data transmission—such as sending a fax as an email attachment. The end system can completely buffer the incoming fax data before displaying or printing the fax image.
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. 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 after learning your physical location.
e911 is another method by which VoIP providers in the US are able to support emergency services. The e911 emergency-calling system automatically associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999 and is being successfully used by many VoIP providers to provide physical address information to emergency service operators.
Integration into global telephone number system
While the wired public switched telephone network (PSTN) 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 and external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider-specific short codes.
Voip phone accessibility and portability
If using a software based soft-phone, calls can only be placed from the computer on which the soft-phone software resides. Thus with a soft-phone the caller is typically limited to a single point of calling. When using a hardware based VoIP phone-device/ phone-adapter it is possible to connect traditional analog phones directly to a VoIP phone-adapter without the need to operate a computer. The converted analog phone signal can then be connected to multiple house phones or extensions, just as any traditional phone company signal can be connected. A second VoIP hardware configuration option involves the use of a specially designed VoIP telephone which incorporates a VoIP phone adapter directly into the phone itself, and which also does not require the use of a computer. A third VoIP hardware configuration option involves the use of a WiFi router and a WiFi SIP phone which can extend a service range throughout a home or office. WiFi SIP phones can also be used at any location where an "unauthenticated" open hotspot Wi-Fi signal is available. However, note that many hotspots require browser-based authentication, which most SIP phones do not support.
Mobile phones & Hand held 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 VoWLAN) or VoIP is implemented over 3G networks. However, "dual mode" telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.
Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network and therefore most VoIP calls from such devices are done over WiFi.
Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.
Another addition to hand held devices are ruggedized bar code 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.
Security
Many consumer VoIP solutions do not support encryption yet, although having a secure phone is much easier to implement with VoIP than traditional phone lines. 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 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.
Pre-paid phone cards
VoIP has become an important technology for phone services to travelers, migrant workers and expatriates, who either, due to not having a fixed or mobile phone or high overseas roaming charges, choose instead to use VoIP services to make their phone calls. Pre-paid phone cards can be used either from a normal phone or from Internet cafes that have phone services. Developing countries and areas with high tourist or immigrant communities generally have a higher uptake.
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 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, potentially making calls appear as though they are from a number that does not belong to the caller. 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.
VoIM
Voice over Instant Messaging (VoIM) presents VoIP as one communication mode among several, with an IM user interface (contact list and presence) as the primary user experience. Many instant messenger services added client-to-client or client-to-PSTN VoIP in the mid-2000s.