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  Fax Features Article

Fax and Commetrex: A 2011 Retrospective

November 29, 2011

We entered 2011 not knowing what it would bring. Well duh, you say. But before you jump on that, let me point out that often you don’t know what’s happened, even when it already has. But for the last few years, the business community has been beset by more economic uncertainty than at any time in the last 70 years. Global finance continues to explore uncharted territory. The only thing we can say about 2011 is it was, apparently, less angst ridden than the previous two years. But now comes 2012, with its uncertainty factor pushed up by problems in the Eurozone. What’s next, horrific economic dislocations caused by the defaults of multiple European nations?


But against this backdrop, business goes on. It seems that progress is inexorable if it can occur in this environment, and occur it does, especially in meeting the prerequisites for the global deployment of an all-IP telecom fabric. Of course, I’m talking about the progress made by the joint efforts of the SIP Forum (News - Alert) and i3 Forum FoIP task groups to prepare real-time FoIP for a global deployment. 

But why does real-time FoIP require special treatment compared with voice? The key reason: unforgiving non-humans at the endpoints implementing a computer-computer protocol with relatively tight timing requirements. Enterprise networks can handle those timing requirements, but international tandem network connections too often do not. To those that say, “If we drag our feet long enough, fax terminals executing the T.30 (G3) fax protocol will just go away…problem solved,” I would say that it’s nearly impossible to overestimate the inertia of a massive installed base. So don’t hold your breath…solve the problem. And that’s what we’re doing at the SIP and i3 Forums.

Of course, there are proprietary ad-hoc solutions, but they come at a high cost, especially when you consider that they are a temporary bridge to the solutions within the framework of the current SIP and T.38 recommendations that will result from the watershed events described here that occurred in 2011. And they are not real-time solutions.

So, what were these “watershed” events in 2011? The short answer is unprecedented testing of the FoIP performance of carrier networks in international call routes. Working together, the two industry groups completed the first phase of our testing program over a 14-week period in the first half of the year. Q3 was spent analyzing the data. From that analysis, the readiness of the global network for international FoIP was characterized (it’s not), many of the problems with carrier-networks were characterized, we’ve come up with several recommendations, and we have determined that further testing is required. 

We are talking about an all-out assault on the problem, and it’s being pushed hard by the carriers. They are serious!

You can read a summary of the Phase I test report on the SIP Forum Website. Some of the observations and recommendations were:

1. Eliminate all non-standard audio attributes in an SDP payload.

2. Consider implementing V.152 for non-T.38 calls (see 10.2 of V.152) (echo cancellation, VAD switch off, and jitter buffer set to fixed value).

3. Eliminate network elements that block RTP media when fax tones are detected.

4. SIP Re-Invites changing an active fax session should be ignored.

5. Network elements in carrier interconnection networks should be transparent to the transmitted payload.

6. Pay attention to UDP (News - Alert) redundancy.

7. PCM integrity was the cause of many failures. We suspect that much is due to peering-point problems. We recommend that this be investigated in Phase II.

8. Re-Invite delay should be as short as possible.

9. T.38 offer in initial Invite, with our without audio, seems to cause risk of failure in interconnection.

10. Consider the use of “user=fax” in an initial SIP invite to inform a carrier’s routing algorithm that the call will likely need a FoIP-qualified route.

Now, in 4Q 2011, Phase II testing is getting underway. Primarily, Phase II is designed to investigate the key problems identified in Phase I. 

But, to look beyond the specific problems identified in the testing to a more general observation, least-cost routing of international calls using SS-7 may not be usable for FoIP routing in its current state. That’s the reason for point 10, above, which gives the routing entity the opportunity to go outside of SS7-based routing to route a fax call, or a suspected fax call, over routes that have been prequalified for FoIP. But just as ENUM, a standard that uses a telephone number to look up a subscriber’s IP addresses and service profile, is required to effectively route an HD-voice call, so too it could be leveraged to overcome the less-than-effective SS7-based routing being used by carriers to route FoIP calls today.

I characterize this testing as a watershed event because, for one thing, it’s unusual for carriers to work directly with what is effectively an equipment-vendor group, especially to test the effectiveness of their network’s use of a particular technology. And, it marks the beginning of the transition from what I call FoIP Phase II to FoIP Phase III. Phase I was the use of FoIP within enterprise networks; Phase II was the extension of FoIP from the enterprise with the VoIP service provider; Phase III is the use of FoIP by the major carriers in international tandem connections. 

So, we just might look back on 2011 and mark it as the year the SIP Forum and the i3 Forum worked together to make FoIP in international carrier networks deployment-ready.


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Edited by Juliana Kenny

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