In my Introducing SIP Trunks article, I started the conversation about moving from TDM to SIP trunks. Today I would like to explore how one goes about choosing the right SIP trunks provider. In the past few years SIP trunks have become widely available with offerings from every major telephony carrier and many of the smaller, regional telephone companies. Still, it is important that an enterprise understand their particular SIP trunks requirements and choose the provider that best meets them. Not all SIP carriers are created equal and each has its own particular strengths and weaknesses.
To help in the decision process, I’ve put together a list of items that should be considered during the analysis of a potential provider.
Reach refers to the availability of SIP trunks from a particular provider in the physical locations your enterprise needs them. For instance, are your needs restricted to a specific city or small geographic region? Are they more expansive than that? Do you have locations throughout the United States in both big and large metropolitan areas? Are you an international company with offices around the world? If you are standardizing on a single carrier it is important that they are able to provide service to all your locations.
You may wish to use the same carrier for both TDM and SIP trunks. This will make it easier to port numbers between the two.
Do you want a carrier that can deliver both your telephony and Internet traffic? Do you have an existing MPLS network and is it necessary that the provider of that network also be the provider of your SIP trunks?
What codecs do you require? Most, if not all, carriers support G.711 and G.729. However, you may also require wideband G.722 or T.38 for fax. If so, ensure that your chosen carrier will support all required codec on both ingress and egress.
Quality of Service
Ask for the Quality of Service (QoS) metrics for the carrier’s SIP trunks. Make certain that your tolerance for jitter, latency, and packet loss can be met. The following metrics are typical of a quality voice call.
• Packet loss of less than 1%
• One way latency (mouth to ear) less than 150ms
• Average, one-way jitter less than 30ms
A carrier may support media encryption (SRTP) as long as they are not required to decrypt it. In other words, the SRTP stream will be sent from the enterprise to another SIP element outside the control of the carrier that will do the eventual decryption. The carrier is simply the conduit for the media and plays no roll in inspecting or manipulating it.
The SIP Refer message is used to move a SIP session from one place to another. Think of that movement as call transfer where a call between Party A and Party B is referred (transferred) to Party C.
A Refer can be used to perform a “take back and transfer” between an enterprise and a SIP trunk provider. When choosing a provider ask if Refer is supported and what special charges might apply when a Refer is processed.
Fax will be supported by the carrier by two different methods. Fax pass-through is the process of sending fax on G.711 media streams. A carrier supports fax pass-through by provisioning particular DID numbers as G.711. T.38 is a Fax over IP (FoIP) protocol that packetizes Fax the T.30 protocol. A carrier supports fax by providing a TDM to T.38 gateway within their network for incoming faxes from the PSTN, a T.38 to TDM gateway for outgoing Faxes to the PSTN, and allows the provisioning of particular DID numbers as T.38.
If you require T.38 on your SIP trunks make sure that the carrier supports it. Not all do.
For a deeper discussion on fax, please refer to my Fax Over IP article.
There are cases when your enterprise requires that certain numbers come in and out of the network using particular codecs. For instance, DID numbers for an IVR must be G.711 while the enterprise uses G.729 for all other calls. T.38 may be required for Fax communication. Ensure that your carrier can provision DID numbers on a block or individual basis.
Traditional TDM Features
If you have existing TDM features that you need to exist on your SIP trunks (e.g. toll free 1-800 service), ensure that your chosen carrier supports them.
I once ran into a situation where the carrier was performing DNIS translation on all incoming calls to a customer’s contact center. Unfortunately, the DNIS service was not available on SIP trunks from that carrier. If we hadn’t caught that during the discovery process the customer’s contact center routing would have failed after the calls were move from their TDM to SIP trunks.
Note that many if not all carriers do not support predictive dialers and on SIP trunks.
Bursting is analogous to check overdraft protection. You write a check for more money than you have in the bank and the bank provides you with a short-term loan. With SIP trunks, bursting means that the carrier will temporarily allow you to exceed the number of your provisioned trunks. Without check overdraft protection, your bad check receives a “not sufficient funds” (NSF) and is rejected. Without bursting, all calls that exceed the provisioned number of trunks receive busy signals.
Different carriers provide bursting in different ways. Some, like Verizon’s BEST program, give you a certain number of trunks over your provisioned amount up to a set maximum. Other carrier simply charge on a per session basis so additional trunks are available until you run out of bandwidth.
How does a carrier charge for trunks? Is it per session or per block of trunks? How do they charge for bursting? Is there a set bandwidth charge? Do they charge for porting a number from the TDM world to the SIP world? How much do they charge to port a number back to TDM?
Learn how a carrier charges for SIP before signing a service agreement.
Reporting and Metrics
What reports does the carrier provide for their trunks? Are they real-time, historical, or both? How are they delivered to your enterprise? What Service Level Agreement (SLA) will the carrier provide?
Interoperability and Support
It is essential that the SIP trunk provider you choose is supported by your SIP communications system. The support question must be asked for your SBC (e.g. Avaya, Sonus, Acme, Audiocodes, etc.) as well as your prime communications system (e.g. Avaya, Siemens, Cisco, etc.). For example, if you plan on using One-Source SIP trunks on an Avaya 6.1 Session Manager with a Sonus 5100 SBC, make certain that all parties consider this a certified configuration.
Perhaps one day when everyone trusts the adaptation provided by SBCs and session management this certification requirement will go away, but until it does it must be accounted for.
Make it Happy
You may have needs beyond what I just described so it’s important to do a thorough analysis of your current trunking configuration prior to moving to SIP. However, please don’t let this process frighten you away from SIP trunks. The positive aspects far outweigh the negatives, but a positive experience will only be achieved through careful planning and roll-out. Do the necessary work up-front and you will be happy that you made the switch.