Asterisk® and VoIP

Welcome to the Asterisk® technology page. From this page, you can learn more about Asterisk, VoIP gateway applications, SIP trunking, and other VoIP-related technologies.

Asterisk Technology Overview
Asterisk is Open Source software that can run on PC hardware to create complete IP PBX solutions that support protocol standards including SIP, H.323, PSTN (public switched telephone network), Skinny, MGCP (Media Gateway Control Protocol) and IAX (Inter-Asterisk eXchange). Asterisk can convert all widely accepted voice signaling protocols, but it is not a SIP proxy server, and it does not provide global routing of SIP calls.

Today, VoIP services such as SIP gateway and SIP trunking are basic requirements for business class routers. For example, a business with employees working from home offices can use the Internet to connect a remote VoIP phone to the corporate SoftPBX or legacy PBX. Similarly, branch offices can be connected to a main office using this approach. As branch offices grow, Asterisk systems can be deployed at branch offices, which can provide local dial tone using FXO ports for analog POTS circuits, or PRI/BRI ports for digital telephone circuits. Once the main and branch offices are interconnected with Asterisk running at each location, call plans can be set up to use the Internet for free calls between branch offices, with local dial tone accessible from any branch.

ImageStream integrated Asterisk into its embedded Linux distribution as an optional package in response to customer demand. SIP and other VoIP gateway services can be deployed without additional hardware if the installation only needs to provide connectivity back to a main office. At the main office and larger branch offices, local dial tone can be provided using optional multi-port FXO, PRI or BRI cards. The call plan determines whether a call is routed over an Internet VPN to a remote extension, over an Internet VPN to a remote switch where local dial tone may be obtained, or via connections to the local telco.

VoIP Codecs
VoIP codecs are used to encode and decode voice traffic that is transported over an IP network. Some codecs are designed to deliver voice using the lowest possible bandwith, which typically provides the lowest call quality. Higher bandwidth codecs normally provide better call quality. Here is a list of codecs supported by Asterisk.
Supported Voice Codecs
  • G.711 ulaw/alaw 64 Kbps Best Quality
  • G.722
  • G.726 ADPCM 32 Kbps High Quality
  • GSM 13 Kbps Good Quality
  • iLBC 15.2 Kbps Good Quality
  • Speex Variable Rate Low to High Quality depending on selected bitrate

Voice Codecs (Pass-through Only)
  • G.723.1 6.3/5.3 Kbps Low Quality
  • G.729 8 Kbps Low (Toll) Quality

Codecs Requiring licensing
  • G.729 8 Kbps Low (Toll) Quality - can be licensed from Sipro Lab Telecom or VoiceAge
Another performance component of these codecs is packet size. With most routers on the market, throughput is directly affected by packet size. As average packet size decreases, total throughput normally falls. As average packet size increases, the router can support higher levels of throughput. ImageStream routers with PCI interfaces behave this way, because bus latency increases as packet size gets smaller. So, when you design a network for VoIP, you not only need to consider the total throughput required to support the application, but you should also consider router performance at the average packet size you will need to support the application.

Quality of Service (QoS)
In a packet switched network, congestion can lead to high latency, and high latency and jitter can bring a VoIP telephone call to an abrupt end. While some VoIP codecs are designed to operate in higher latency environments, ITU-T G.114 recommends a latency of no more than 150 ms for VoIP calls. Jitter is the variation of latency over time, so the best routers will have both low latency and low jitter.

Once a packet leaves the network, there are no guarantees of low-latency delivery via the Internet. Some Tier 1 ISPs offer service level agreements (SLAs) that specify maximum latency and jitter. Different networking technologies may also include some level QoS to ensure low-latency delivery of VoIP packets. For example, at the link layer, a high-overhead technology like ATM can be used to ensure low-latency delivery of voice packets across the ATM network. Lower overhead Layer 2 link technologies like ethernet can also be used in conjunction with QoS to ensure reliable delivery of high-priority packets throughut the networks you control.

Under all circumstances, the latency of your upstream provider and other hops on the Internet can become irrelevant to VoIP performance if you can't get packets out of your network at the required priority level. The faster packets move through your network, the lower your first hop latency will be, and the more likely your VoIP calls will not drop out. To a certain degree, you will only be able to control your own network, and you may be able to ensure your upstream provider will deliver VoIP traffic at an acceptable latency and jitter through an SLA. These limits make proper QoS design and deployment a requirement for highly reliable VoIP networks.

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