While SIP Voice over IP (VoIP) technology has come a long way to improve connectivity and quality of service, there remains numerous potential points of failure along the path of devices over which the data flows. Unfortunately, if a remote location is experiencing an issue with their data connections, not only won’t they be able to call support to troubleshoot the issue, but they may not be able to send an email either, as both lines of communication rely on the same network. This is why it’s important to proactively monitor your Session Initiation Protocol (SIP) network to detect issues as soon as communication is lost. To begin to understand the complexity of the SIP communication protocol, let’s take a look at the path required to maintain a successful SIP connection.
Most VoIP systems reside in a data center behind several layers of devices that include routers, firewalls, and modems. Each of these devices maintains complex algorithms that determine how to handle all Internet traffic, including your sensitive VoIP packets. Once your voice traffic has navigated your internal environment, it encounters the external/public Internet on the data side, and the public switched telephone network (PSTN) on the voice side.
For example, let’s look at the path of a sales representative working out of their home office in Minneapolis, MN and connecting to the VoIP server at the headquarters in Austin, TX. The sales rep plugs their SIP-based phone into their local network. The phone then sends traffic from their router and through their modem where it must perform a DNS query to figure out how to find the home office.
The modem checks with the local Minneapolis ISP DNS servers and queries the top-level DNS servers, which sends them to the Authoritative DNS server for your company’s domain. Hopefully, each primary DNS server along the path responds efficiently, or your DNS resolution may be delayed while the process continues to query for alternate routes through additional DNS servers.
Maintaining Connection Through SIP Registration
Once the SIP phone has established a connection to the main office and navigated the internal network that houses your SIP servers, the sales representative will receive a dial tone and can place a call. The SIP protocol typically keeps the session between the phone and the server open by transmitting SIP registration to the server on a regular basis using UDP. SIP registration verifies with the server what the IP address of the SIP device is and that it is available to establish or receive a phone call.
Establishing a Connection to the Recipient
When the user dials a number, the SIP phone does not know how to reach the final destination, so it sends an invite request to the SIP server. The SIP server will use the E. 164 number mapping standard (ENUM), translating the telephone number into a Uniform Resource Identifier (URI), and determining the path to the phone number that has been dialed. Once the server has received a successful Naming Authority Pointer Resource (NAPTR) from the DNS authority, the VoIP server then reaches out to the public switched telephone network (PSTN) over the Internet. From here, the traffic may travel across numerous different data networks to reach the intended recipient. Now that your user has finally established a connection, they are able to complete the call.
Monitoring SIP Connectivity
As you can see, every step along this path contains multiple potential points of failure. Even if you have internal monitoring on your servers, are you able to continually test your systems for connectivity from outside your network from locations around the world? To ensure that your systems are available all the time, and to receive notifications if there is ever a problem with your systems, sign up for a free 30-day trial to monitor your SIP activity now.