Solving VoIP Component Connectivity Problems with SIP Monitoring
VoIP and Service Quality Overview
Businesses are increasingly using Voice over Internal Protocol (VoIP) for their phone service1. As such, the negative impacts on business performance, profitability, and revenue when VoIP services experience degradation or downtime is also increasing.
While VoIP telephony systems continue to gain market share and have become the primary voice communication system for many businesses, they also have some inherent challenges compared to traditional public switched telephone network (PSTN) services. For instance, while carrier-call PSTN services are typically high-quality and low-latency, VoIP services are bandwidth and delay sensitive. Additionally, successful completion of VoIP calls usually depends on multiple components working properly that can be outside of company control.
Session Initiation Protocol (SIP) Monitoring
Dotcom-Monitor® SIP Monitoring is an online monitoring service that proactively monitors the ability of VoIP infrastructure components to establish and maintain VoIP calls. It proactively monitors VoIP services using Session Initiation Protocol (SIP) the signaling protocol typically used for VoIP. The SIP-based monitoring service acts like an end client, by periodically placing VoIP telephone calls to a specified number and checking the results of that call.
To accomplish this, SIP Monitoring is provisioned as either an extension, or a client, on the VoIP system and configured to call a specific number using a specified SIP server with certain parameters. The expected result of the call is setup as "Answer", "No Answer", "Busy", or an Error Condition (if there is an unexpected result).
The service could be used in multiple ways to constantly test VoIP network availability. Here are just few examples:
- A business running a VoIP PBX, for example such as Asterisk, can setup SIP-based monitoring to periodically call a voice-mail number and check the expected result that the call is answered by the voice-mail.
- A VoIP Service Provider can setup proactive SIP-based monitoring. Each customer could be setup with a special "test" number that the Dotcom-Monitor® service would dial. The expected outcome to the number would be set to "Answer" or "No Answer" depending on whether the call goes into voice mail.
- A VoIP Wholesaler can proactively monitor specific route destinations by periodically dialing external numbers routed through their partners to ensure calls to those destinations are properly delivered.
VoIP Deployment Challenges and SIP-based Monitoring
Because VoIP systems are generally distributed, heterogeneous systems that include multiple components and depend on numerous third-party providers to function properly, there are more potential points of service failure. When a VoIP system service failure does occur a proactive SIP-based monitoring system discovers the error immediately and the resulting error report helps to pinpoint where the error condition is occurring.
For example, it is typical for a VoIP system to have one or more servers that act as a hub for managing VoIP traffic. These servers handle incoming and outgoing calls, voice-mail, call queuing, conferencing, and other components. Additionally, these VoIP servers manage the handsets (telephones) that end-users use to place and receive calls. Each handset is an IP-enabled device with a distinct IP address. In this example, to ensure that each component - servers, handsets, and routers - in the VoIP system is operating properly, a proactive SIP-based monitoring system monitors each VoIP component’s ability to establish and/or maintain the following:
- Outgoing call succession
- Inbound call routing
- Internal extension connectivity
- Voice-mail availability
SIP Monitoring Setup Steps
Setting up a SIP task with Dotcom-Monitor®â€™s SIP Monitoring tool is a simple process. First, provision a SIP Monitoring extension on the telephony systems. Next, add a new device (for example "Asterisk") to the Dotcom-Monitor® control panel. Then, specify the "Asterisk" device as a "SIP" task. Finally, set the target and parameters of the new task of "SIP" type, (see picture below and field descriptions that follow):

- Task Name– Name that identifies the task, for example "Asterisk".
- Maximum Connection Timeout (in Seconds) -Time to wait for an expected call result.
- Server – Name of VoIP server to be monitored. For example: asterisk.yourcompany.com
- Port – This is an optional field. If not specified, Dotcom-Monitor® uses default SIP Port 5060
- User Name/Password – Credentials used to authenticate within the server. For example: 209 – username and "password". The task does not keep state. SIP Monitoring will login into the system, authenticate, place a phone call, and logout. Dotcom-Monitor® does NOT register with the service (instead, SIP Monitoring makes a temporary connection, conducts a test, and then disconnects).
- ANI - This is an optional field. This is a number that used as caller ID. It can be set for some systems to accomplish custom routing for calls generated from SIP Monitoring. For example, the system can be programmed to dial a specific number and if "ANI=209" set a busy tone.
- Phone to Dial – Number to dial. This can be any number as long as the telephony system knows how to route it.
- Expected Call Result - If a call is successful and did not produce any SIP communication errors, then SIP Monitoring analyzes the result of the call. The analysis can be setup as "No Answer", "Answer", or "Busy" depending on the expected outcome.
SIP Monitoring Error Detection
Error detection occurs at each step of the SIP Monitoring process. If an error is detected, SIP Monitoring records all properties of the error, which helps to pinpoint where the error condition is occurring. SIP Monitoring reports these error properties as Blobs of text accessible in online reports:
SIP Final Response:408
SIP Final Response Description: Request Timeout
Media: False
TimedOut: True
VoIP Monitoring System Tools
After SIP Monitoring replicates a call to an end-user’s SIP device it then analyzes the call responses to determine the connectivity of VoIP system services. When a problem is detected, the SIP Monitoring notification feature sends an alert via phone, pager, email, or SMS. Additionally, real-time connectivity status reports are available via an intuitive online Dashboard interface with enough detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for VoIP management purposes, including Service Level Agreement (SLA) issues.
1VOIP News, State of the VIOP Market 2008 http://www.voip-news.com/feature/state-voip-market-2008-031008/ |