I often hear questions from customers about SIP. SIP can mean different things to different people. SIP trunking, end to end SIP, SIP for voice and video, SIP in the contact center –those are just a few of the variations that come to mind. For purpose of this post and others to follow, as well as my upcoming presentation at International Avaya Users Group (IAUG, where I’ll be speaking next month in Orlando, FL) I am going with end to end SIP for voice as a starting point.
When I talk about end to end SIP I am looking at SIP all the way in from the carrier trunks, through the core infrastructure, and out to the phones (hard phones or soft phones). With that out of the way, let’s talk more about some of the questions that need to be answered and some of the key things that you, the customer, need to do in order to make your SIP deployment successful.
First, there are a few core design choices to consider up front to ensure long term goals are met. For instance, when looking at the design of my network, do I want to put SIP trunks from a managed service provider directly to each site or SIP trunks into a core set of data centers and then drive traffic out to my remote sites via a MPLS network? The answer to this partially is determined by size of both the organization and the sites that need to be supported.
Another consideration in answering this question is what level of autonomy each site can handle from a IT/Telecom perspective, but also from a failover perspective. Obviously SIP directly to each site provides more autonomy, but then you may not get some of the economies of scale and support that a centralized SIP based solution can offer. Unless you go with redundant suppliers to each site you still have a single point of failure, but then much of the cost savings that you were hoping for may be gone. I typically see a more centralized SIP deployment with redundant sites and carriers to each site.
The next set of choices deals with the now and the future: do we want to have a converged network for voice, data, video, etc.? This needs to be thought of ahead of time even if this is not the goal of the first stage of your deployment. You will want to design and choose vendors that support your end vision.
Related to this is how you handle codec choices where bandwidth and quality tradeoffs occur. Even if the network is just “voice,” you need to make the choice between G.711 (“toll quality”), G.729 (compressed), G.722 (“HD voice”). When making this choice, the thing most people look at the tradeoff between quality and bandwidth. However, there are other considerations to take into account, like fax and how it will behave over your IP network (that is if people still use faxes in your company).
One other major component you need to understand is E911 for emergency calls. Understanding how it is handled by your carriers and your vendors, and how the location of the caller affects how the call makes it to the appropriate PSAP are key elements to consider when designing your solution.
Those are some of the core concepts and questions I usually take into consideration. Please post your thoughts on things you think I missed in the comments below.
Coming soon, I will talk about top tips for a successful SIP implementation. Keep an eye out for it!
For more information, learn about Getting the Most Out of Your Migration to SIP Trunking.