SIP Trunking Featured Article

August 11, 2008

SIP Tips -- 'Smart Devices - Dumb Networks'

By Tom Cross
Technology Columnist

Podcast Series on Selling SIP Systems & Solutions - “You Can’t Sell What You Don’t Know”

I just taught a class to new hires to the SIP industry and often got asked the question, “What is SIP?” One answer that I increasingly give is, “in the old days, phones were dumb like POTS and networks were smart like the PSTN. In the new Internet world, phones are smart and networks are dumb.” Check the animation here to see what I am talking about:

If you prefer a more technical tutorial, here goes. With SIP Trunking, the IP media stream coming from within the enterprise stays as an IP media stream and passes to anywhere within the enterprise or across the boundary of the enterprise to another enterprise via IP. This reduces the need for local telephone systems using instead hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) producing considerable savings. In addition, considerable savings are also found by eliminating expensive telephone desksets by using intelligent softphones.
In addition, if you want to sell new SIP solutions, you need to know a lot more. At a recent conference, the panelists on the session Selling SIP Trunking missed the mark in helping the SRO audience get to square one for selling SIP Trunking.  Without going into all the issues they could have and should have addressed, here are three tips. 
1 - What is SIP?  Make sure you know what SIP means.  It means Session Initiation Protocol, not anything else.  Basically, SIP provides signaling, like car traffic lights, in order for SIP devices to call other SIP devices over a symmetrical broadband Internet connection (no ADSL).  If you want to know more about the working of SIP protocol, get involved in technical discussions or your product interoperability compliant, go to the SIP Forum, a nonprofit industry interoperability organization at The SIP Forum will help you understand the industry, players, protocols such as RTP-Realtime Transport Protocol, SDP-Session Description Protocol and others as well as RFC-Request For Comments that are the basis for all SIP development.
2 – SIP devices can be hardphones, wireless phones, softphones (software) and other devices such as soda machines and in the future nearly every other device.  SIP moves the “intelligence” from the PBX/CO into the device.  That is, SIP devices communicate directly with one another without the need for a PBX or CO (Central Office) switching system.  This is similar to the way your PC communicates directly with a Web site. This means features are in the SIP device, not PBX.  For example, I can use my laptop with softphone software as a telephone and can take it anywhere and plug in to an Internet connection and begin making outgoing or receiving incoming calls from other SIP devices without a PBX.  If I need to call outside my SIP network or receive a call, my SIP gateway provider (in this case gives me a PSTN number which can be used no matter where I am.  Features such as voice mail, transfer, conference, etc. can be added through software from the SIP system or SIP gateway provider.
3 – Bandwidth planning is paramount.  SIP devices use a CODEC (coder-decoder, compression-decompression), a technical term for computer chip, to process calls into international standard voice formats.  One major CODEC is G.711 provides for high-performance “toll-quality” calls and uses 64 KBPS per call.  A low-performance CODEC (much like cellular service) for low-bandwidth voice calls of 8 KBPS is G.729.  There are other CODECS supported by various manufacturers. Check specific companies for details. 
The most important point is that in planning for SIP implementations allocate 80-100 KBPS per call for G.711 and around 30 KBPS per call for G.729.  That is, while G.711 uses 64KBPS of voice it needs more bandwidth because of the packetizing (RTP-TCP/UDP-IP overhead) for an Internet protocol network.  Here’s an easy rule of thumb, for G.711 take the total number of simultaneous (concurrent) calls times 100 KBPS and that is the bandwidth the customer needs for peak “busy hour” times.  In addition, SIP trunking providers will limit the number of voice calls based on the CODECS they support.   One SIP trunking provider supports 11 calls using G.711 and 42 calls with G.729. However, the customer benefits when users are not on the telephone with the bandwidth automatically or “dynamically” available for data needs. In other words, check with your SIP trunking provider, media gateway manufacturer and other “parts” in the network. That is, YMMV-your mileage may certainly vary. 
There are dozens of other critical concepts such as security, interoperability, pre-installation planning, data systems integration and others you need to be “SIP smart” in selling SIP that are included in OCS-101 and SIP Planning Guide 2.5 available in the onsite and online courses. 
Discounts are also available to members of the SIP Forum and MS Partners for $99 per student. For customizing, special discounts, Web site animations, technical/sales training, technical writing and other services, go to or please call Tom Cross at 303-594-1694 or
This is also included in a TMC University special course on Microsoft OCS at Internet Telephony Conference & EXPO (ITEXPO) in Los Angeles, September 16-18, 2008.

Tom Cross, who has three decades of startup and consulting experience, writes the CrossTalk column for TMCnet. To read more of Tom’s articles, please visit his columnist page.
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