By TMCnet Special Guest
Brad Johnson, Lead developer for Ecessa
Building Redundancy and Failover into SIP-Enabled Networks
Communications has always been the foundation for commerce, and today businesses have a variety of communication options. The most powerful form of communications is still face-to-face meetings, but sometimes it is not feasible in terms of cost and logistics for businesses. . Electronic communication via email and instant messaging are the most convenient and cost-effective, but lack intimacy.
Yet, despite all the technological advances in communications, the greatest balance of efficiency and intimacy is still the telephone, and this explains why businesses place so much importance on their voice networks. However, the many issues involved in managing voice networks make obtaining reliability more difficult than ever.
Central to this is the fact that today’s IP-based networks are more complex. In addition to legacy POTS, companies now have VoIP and wireless voice networks. Straightforward voicemail has evolved into complicated integrated messaging platforms, and the traditional handset has been replaced by devices such as soft (IP) phones, cellular phones and smart phones. When it comes to voice networks, the demands for availability are higher than ever before. Businesses require networks to work without interruptions and the cost of losing phone services has such a devastating effect on organizations it requires five nines of reliability.
Today, one of the most critical issues businesses face is how to effectively manage voice networks, and the potential for integrating voice systems with multimedia and new interactive capabilities. It is within this challenge that there lies an opportunity for service providers, system integrators, VARs equipped with the appropriate solutions to create new revenue-generating services.
The Growing Deployment of SIP Communications
The age of disparate voice and data networks is coming to an end. Single, converged networks are beginning to deliver all forms of communications and in addition to reducing costs, this convergence can enable a great diversity of new services. At the center of this transformation is Session Initiation Protocol (News - Alert) (SIP), an IP telephony standard developed to manage the setup and tear-down of voice-related IP networking sessions. As voice, data, and video communications become more unified, IP-based applications will increasingly use SIP to manage the connections.
SIP enables communication devices such as fixed and mobile phones to interoperate with Internet services such as email, the web, instant messaging, and multimedia collaboration. Beyond VoIP, SIP-based applications can include many innovative services, such as video conferencing, streaming media distribution, online gaming, voice-enriched e-commerce, and other services.
With the increasing deployment of SIP technology, telco’s and service providers need to deliver five nines of service availability and quality application delivery. Similarly, the need for carrier-grade service requirements has placed the same demands on equipment providers and the VARs and system integrators charged with deploying these solutions.
When it comes to delivering high-availability and performance needed for applications that use SIP, fundamental problems that require network redundancy and failover need to be addressed to ensure calls remain connected, even in the case of a service provider outage. Specifically, problems with NATing a SIP-enabled network must be tackled to guarantee true redundancy for SIP traffic going over multiple WAN links.
SIP and Network Address Translation (NAT)
There are several areas within a SIP-enabled network where NAT and Remote Transport Protocol (RTP) become a problem. SIP packets go from users (with NAT) from their private (un-routable) IP addresses coded into the message headers. However, the SIP packets may not be processed by a NAT device over the WAN since NATing operates only on IP packets. Therefore, when the SIP packets arrive at its destination, they are processed and responded to with unusable source address information. Responses to requests cannot be routed back to the originating party, as the addressing information is not globally routable.
Additionally, Session Description Protocol (SDP) messages used to negotiate the session format (codecs, ports, IP's, etc.), are often enclosed within the body of the SIP message. However, SDP messages are not processed by a SIP proxy (according to IETF standards). Therefore, they will contain non-routable contact information.
Specialized WAN link controllers can address the issues having to do with the NATing of SIP traffic – enabling SIP traffic to go over a NAT environment successfully. The ability to provide failover and load balance SIP connections over multiple WAN links is critical for successful, scalable VoIP environments.
This diagram shows a WAN link controller managing SIP traffic for both LAN and WAN traffic
The Need for SIP-Enabled WAN Link Controllers
A SIP-enabled WAN link controller functions as a NAT device and SIP Proxy within the network. It provides a standard approach to ensure reliable SIP service availability across multiple and diverse WAN connections. While non-SIP-enabled WAN link solutions may be helpful in solving general link load balancing, there are inherent problems associated with failure in supporting SIP message flow, and interoperability issues associated with NATing SIP traffic.
A SIP-enabled WAN link controller is required to perform SIP operations such as session management and routing, transport conversion and security functions. By acting as a NAT device and SIP Proxy the SIP-enabled WAN link controller can perform these operations, and avoid NATing interoperability issues. A SIP-enabled WAN link controller functions as a default gateway for the traffic to be routed through it. This enables the routing of all traffic through the WAN link controller regardless of SIP signaling, media or other types of traffic. A SIP-enabled WAN link controller can be configured as an inbound Proxy for all traffic appointed by the DNS servers, or peering SIP elements, as well as the outbound Proxy for all traffic going out from the LAN.
SIP-enabled WAN Link Controllers Deliver WAN Link Redundancy and Failover for SIP Deployments
In much the same way as dedicated global load balancers provided high-availability, scalability and performance for web applications in the late 1990’s, SIP-enabled WAN link controllers are delivering the same ability to scale, with high-availability and reliable communications for VoIP and other applications that are SIP-enabled.
For enterprises, telcos, service providers, VARs and system integrators, SIP-enabled WAN link controllers can dramatically simplify and reduce SIP application deployment time. A SIP-enabled WAN link controller should provide the following capabilities:
- Deliver seamless call failover for inbound and outbound connectivity
- Enable SIP devices to work over multiple, diverse WAN connections
- Eliminate problems associated with NAT
- Provide link load balancing for SIP traffic
Summary
SIP deployments are continuing to increase, while due to their mission-critical nature, availability, scalability and security requirements have likewise become essential for their deployment. This advancement has resulted in the need for SIP-enabled WAN link controllers that can accommodate carrier-grade operational requirements within a single, purpose-built device. Using a SIP-enabled WAN link controller that functions as a NAT device, SIP Proxy and SIP registrar within the network will ensure SIP service availability, while providing an easy-to-deploy, manage and monitor solution for reliable service delivery. While server and global load balancers are appropriate for load balancing servers and dispersed sites, they do not provide the ability to create the automated failover and redundant WAN connectivity to support reliable SIP network connectivity.
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Edited by Stefania Viscusi
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