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SIP Did for VoIPThe TAM100-VoIP, T1/E1 VoIP Network/Gateway Analyzer provides VoIP network users and service providers a low cost tool to test, verify and maintain VoIP networks at the WAN interface point for gateways, IP-PBX/switches and medium capacity legacy network terminals.
Our popular VoIP Tracer Pack combined with the SAFIRE Professional development environment; the perfect combination for developing signaling applications, then validating & observing the application signaling together with Internet & VoIP signaling! |
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| History Voice over Internet Protocol has been a subject of interest almost since the first computer network. By 1973, voice was being transmitted over the early Internet.[1] The technology for transmitting voice conversations over the internet has been available to end users since at least the 1990's. In 1996, a shrink-wrapped software product called Vocaltec Internet Phone Release 4 provided VoIP, along with extra features such as voice mail and caller id. However, it did not offer a gateway to the analog POTS, so it was only possible to speak to other Vocaltec Internet Phone users.[2] In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace a traditional hardware switchboards by serving as the gateway between two telephone networks. Functionality VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive using the more traditional PSTN. Examples include: * The ability to transmit more than one telephone call down the same broadband-connected telephone line. This can make VoIP a simple way to add an extra telephone line to a home or office. * 3-way calling, call forwarding, automatic redial, and caller ID; features that traditional telecommunication companies (telcos) normally charge extra for. * Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone over traditional phone lines, like digitizing and digital transmission are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream. * Location independence. Only an internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection. * Integration 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. 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.[9] 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. Technical details The two major competing standards for VoIP are the ITU standard H.323 and the IETF standard SIP. Initially H.323 was the most popular protocol, though in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage. However, in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones[citation needed], and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP. Where VoIP travels through multiple providers' softswitches the concepts of Full Media Proxy and Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2) H.245; 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically, to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimization techniques used, such as silence suppression and header compression. This can typically save 35% on bandwidth usage. VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by multiplexing multiple conversations that are heading to the same destination and wrapping them up inside the same packets. Because the packet header overhead is shared between many simultaneous streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only about 1 kbit/s. |
