Setting up Your Service: The SIP RTP Connection Explained
Even though statistics show that people make an average of 5 phone calls a day, you’re probably not thinking about what your phone has to do to make that communication easy and successful.
Important protocols involved in calling your grandmother to tell her, “hello”, include SIP RTP.
When setting up your service, it’s important to start with these basics. The SIP RTP relationship is important because the two can not function properly without each other.
Let’s break down their roles and see how their connection plays a part in your everyday life.
SIP stands for Session Initiation Protocol.
This is the first step in setting up phone calls, as it’s the signaling phase. The signal goes out over an IP (Internet Protocol) network, to IP devices.
Before you even speak a word to the person on the other end of the line or view anything on a web page, SIP has already done an important job.
SIP sends an invitation and receives confirmation from the two end servers who wish to connect and communicate.
Here’s an easy comparison
It works almost like the technology version of seeing someone you know at the grocery store. Think of the sound waves from talking as the signals coming from SIP.
You see each other from afar and your brain tells you who the person is, how you know them, what their name is, etc. This is SIP registering two end servers that want to connect.
Then, you smile, nod, or walk closer to the person to signal that you want to initiate a conversation. This is SIP inviting the two end servers to connect and waiting for their confirmation.
Finally, you are connected and here is where we can insert RTP (the voice and audio session where you are chatting and catching up with this old friend).
Disconnect (walk away) and both SIP RTP jobs are complete.
RTP stands for Real-time Transport Protocol.
After you punch in someone’s phone number and SIP gets a confirmation for both end servers to connect, you don’t want to sit in silence or not be able to communicate.
RTP sends real-time data through the two connected end systems that SIP put together. This means the talking and viewing aspect of your task.
RTP works with many other protocols and services to make sure your phone conversation runs as smoothly as possible.
As you’ll read below in the troubleshooting section, RTP is necessary because it self-monitors itself to detect any possible shortcomings that might arise.
Companies can benefit from SIP trunking because it is cost effective for those who deal with high volumes of business/communication, especially long distance.
SIP trunks do not just aid in voice calls, they also provide an enhanced 9-1-1 call service, send instant text messages, and let you have conference calls through various media outlets (phone, computer, etc.)
The “trunk” is a physical line that helps connect servers (as opposed to virtual connections). Like stated above, companies prefer to SIP trunk because the physical line has less room for error.
When you’re trying to make a business call with a partner halfway across the world, it’s not a good idea to chance the connection with a virtual line, so trunking saves the day.
SIP trunking is also convenient and beneficial for your company because it makes it easy to transfer calls quickly and efficiently. Less wait time for customers makes for happier customers.
Setting up your service
Companies like SipTrunk.com actually allow you to take control of how many connections you are making at the same time.
When you think about setting up your own service, it’s important to know what kind of connections you’re looking for and what kind of security you have/need.
If you choose SIP trunking, what the signal travels on (physically) differs from if you were not SIP trunking.
The signal carries across an ISDN (Integrated Services Digital Network) line, or a copper line. These two options make it possible for you to be taking an average of 30 or more calls at a time, increasing your business.
We keep talking about the SIP RTP relationship, but it’s actually a good thing that the two are separate entities.
When troubleshooting voice calls, the two can be tested separately to identify the problem.
Usually, when calls are being dropped from your phone and neither side of the call can register authority, it’s a network issue.
If the network is tested and there seems to be no congestion or issues, it may be a hardware problem for the phone.
Possible hardware problems
SIP’s data structure has information that is easier to read in its code form because it’s a text based protocol. It also has its own error codes, which makes problems even easier to identify.
Error codes include
- 404 (not found)
- 408 (timeout)
RTP’s data structure is a little more complicated to read.
Since RTP is happening in real time, it monitors itself to constantly (through RTCP) be checking for lags, jitter, and “packet loss”.
RTCP is a protocol that analyzes the data coming from the running RTP. They work hand in hand to avoid issues with the audio/visual streaming.
Once you can identify which entity is giving you an issue, you can work to troubleshoot it specifically and solve your problem.
The SIP RTP Connection
Simply, the SIP RTP relationship can be broken down into sections.
- SIP recognizes two servers that want to connect
- SIP registers the servers and invites them to connect
- The servers are connected and can be disconnected
- RTP comes once the connection is in place and audio/visual communication can begin
It is apparent that you can not have a successful phone conversation, video chat session, or browse the internet without the SIP RTP relationship running smoothly.
Now that you know how the two are connected and work together, you’re ready to get started setting up your own service.
When you look at how SIP trunking can benefit your company you will see how it saves you time and money.