SIP-Powered Phone Calls Explained | Flowroute Blog

SIP-Powered Phone Calls Explained

Posted on July 28, 2020

Although industry naysayers claimed voice was dead, phone calls remain an integral part of enterprise and end user communication strategies. And while the platforms we use to communicate have evolved (i.e. smart phones or web browser-based conference tools, etc.), some of the foundational elements of calling have remained the same. For example, every phone call consists of two components: signaling and media.

Signaling is the foundation of phone calls and does the work of establishing, maintaining and ending the call. Media is the actual call audio. On a VoIP connection, media is separated into digital packets for efficient transfer between endpoints (i.e. callers and phones). There are a few different signaling protocols that go into calling, but for this post we will focus on Session Initiation Protocols – also known as SIP.

While we are looking at the foundational elements of calling, it is important to know the components of traditional phone systems. There are PBX (Private Branch Exchange) systems, which are on-premises systems that manages calls; PRI (Primary Rate Interface) lines, which are the lines that connect calls to the PSTN; and lastly, PSTN or (Public Switched Telephone Network), which is the network that routes calls to their destination PBX.

SIP calling removes the need for PRI lines and is the process of transmitting voice calls over a SIP trunk or a SIP channel. A SIP trunk is virtually installed over your internet connection and connects the PBX to the PSTN, bypassing the PRI lines. The results of SIP capabilities include call management features like auto attendants, call forwarding, voicemail and much more without requiring analog or multiple phone lines.

Many companies rely on SIP calling because it is cost-effective, scalable, flexible and reliable. But how does it work?

When it comes to SIP calling, signaling and media play vital roles. Signaling starts the SIP call by completing three key jobs.

Job #1: Send the INVITE

When you dial a number, your phone system sends a SIP packet to your carrier (see a sample SIP packet here). That SIP packet contains all the data necessary to create the call to your new prospect.

The SIP packet, which is responsible for creating the call, is known as the INVITE. Your carrier uses the INVITE as a notification of an intended call. It then performs a quick LRN (Location Routing Number) lookup, if applicable, to locate the number that you requested in the “Request” portion of the SIP packet. The LRN procedure is critical for NANPA (North American Numbering Plan Administrator) phone numbers in today’s day and age because a number port order can be completed more quickly than ever before.  Without completing the LRN process, your phone call could be sent to the carrier that originally hosted the number, even if the number has since been ported with a different carrier. If that happens, the destination caller will likely not be the intended receiver.

Within about half a second your carrier has figured out where your call needs to go and has sent your SIP packet, INVITE and all, to the number you dialed.

Job 2: Arrive at the destination

When your INVITE reaches the destination carrier, that carrier typically sends back a provisional response (which in SIP is ‘1xx’) which means, “Wait, I’m looking for that number on my network, and while I do you won’t be billed.”

When your INVITE arrives at its destination, a few lines of information riding along with your SIP packet called “Session Description Protocol” or SDP, facilitates the introduction in terms of how the media or “meat” of the call should be set up. This includes the media ports to use and the audio codecs the sender is partial to.

Job 3: Establish call parameters

When your call is answered, either by a person, voicemail or Interactive Voice Response (IVR) robot, a ‘200’ response is sent back to your system indicating that your call has been received. The response also carries additional SDP parameters outlining, “Here’s how I’m willing to talk” to finalize negotiations of the call.

The calling party sends back an acknowledgement (‘ACK’) that they’ve received the ‘200’. This also ensures the call prevails and is not dropped mid-conversation. In a matter of one second (not including how long it took for the other end to pick up the phone), the call is established.

After signaling, media (aka call audio) takes over. Once signaling has completed its job, media flows between the established ports as a series of digital packets. The focus then shifts to call quality.

Phone calls are a real time communication medium. Unlike other IP-based connections that transport information in a series of bite sized packets, if an audio packet gets dropped because of a temporarily lost connection or lag, it cannot be resent. Resending packets will deliver them out of order, which equates to your words or syllables coming out scrambled.

To boost call quality, stay away from internet service aggregators that resell services from several providers based on lowest price. You can also cut out a transmission step by dealing directly with the end provider.

While media transmits, SIP checks in periodically to make sure the call is still happening.

These check-ins are called “session timers.” You can set the length of session timer in your phone system programming. Session timers are important because they make sure calls are ended in the event of any unexpected hiccups. For example, without session timers, a call could become unresponsive and unable to correctly receive or end calls.

Session times are set by sending a “REINVITE” at pre-programmed intervals. If the call is still on, the other end sends back ‘100’, to which the caller responds with ‘200’, to which the callee responds with ‘ACK’. If any of those legs don’t happen, the SIP tells your phone system the transaction has failed, at which point, your system MUST send a ‘BYE’ to officially end the call dialogue.

Understanding the subtle technical complexities of SIP calling can help businesses gain insight into an essential business communication function that impacts their bottom line. Further, ensuring robust call quality and calling capabilities have the potential to improve customer retention and employee communication. Knowing how signally and media operate behind the scenes is a good first step to actualizing those benefits and taking control of your communication offerings.

Flowroute’s inbound SIP trunking provides unlimited concurrent call capacity, with no limitations or restrictions. As your volume increases, new instances are dynamically created to help you scale your voice services. We also deliver outbound call audio over the shortest path possible to increase call quality and lower call costs locally and internationally. Find out how Flowroute can support your SIP calling needs.

 

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