I don’t think anyone questions the fact that most of enterprise communications is going SIP. We hear it everywhere. But I still get people asking about why it is such a big deal. Isn’t it just another signaling protocol? Is there really a strategic advantage to go SIP? The answer is an overwhelming YES! SIP’s flexibility and interoperability makes it an obvious choice. But I do think it’s important to dive into the details to know exactly why SIP is such a big deal. I think of SIP having three major strategic use cases. Generally in order of adoption, they are:
- Adjunct connectivity and routing,
- Trunking, and
- Endpoints.
What’s interesting about these use cases is that there are almost always boring ways to implement and migrate to SIP, and there are also very cool, game-changing ways to migrate to SIP. Let’s talk about it all!
An Introduction to SIP
I would first like to level set and remind people that SIP only describes the signaling of a session. It is not responsible for actually delivering the session itself. In old telephony terms, we’d think of the signaling path and the bearer path. Just like an ISDN-PRI trunk has a signaling channel (i.e., D-Channel) and bearer channels (i.e., trunk members), voice over IP (VoIP) does the same thing. You have codecs that help facilitate the voice portion, and you have some kind of signaling channel. The first mainstream signaling protocol for VoIP was H.323. If you dig into the H.323 protocol, you will find that Q.931 is a key component. Not sure what Q.931 is? That’s the same protocol that defines the makeup of an ISDN-PRI D Channel. So, in essence, H.323 is just a packetized version of an ISDN-PRI D Channel. Cool, huh? Yep. But at the same time, that’s the biggest limitation as well. H.323 is only really good at signaling for voice!
SIP, as a direct replacement to H.323, can signal for almost anything. Can it signal for voice? Yes! Video? Yes! Presence? Yes! CTI screen pops? Yes! Multiplayer chess pieces? Yes! Both of these signaling protocols (H.323 and SIP) can be used for today’s VoIP. Generally speaking, the existing ways we deliver VoIP Bearer paths (MedPros, Gateway DSPs, Media Servers, etc.) are exactly the same, regardless of the signaling method. Now, moving on to the use cases!
Adjunct Connectivity + Routing
As I mentioned, adjunct connectivity and routing is generally what gets adopted first. Mostly because you don't have a choice. There are very few manufacturers that want to create TDM-oriented adjuncts (think voicemail, conference bridges, IVRs, etc.) anymore. Prior to SIP, the only truly vendor-agnostic connectivity was with analog ports or T1s. Both of these required physical ports per connection. It was very hardware-intensive, and impossible to virtualize.
SIP has the benefit of being universally accepted for basic integration (often referred to as the SIPPING-19), as well as having the extensible capability of being able to be expanded to support very robust features (even vendor-proprietary features). So, most of the major vendors have converted their products over to SIP. Most customers’ first entrance into SIP was some type of session manager to help facilitate the connectivity between the core system and these adjunctions. Soon, people started realizing that this same SIP session management core could be used to connect different systems together and different locations together. Customers started creating their own SIP-based service provider cloud within their enterprise, allowing them to share redundant resources between locations. This allowed for huge cost reductions and much more easily deployed applications across their entire enterprise.
Continue on to the second part of this blog series, where I dive into the uses cases for SIP trunking and endpoints!
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Topics: Unified Communications
