Archive

Archive for June, 2011

Four Use Cases for Monetizing Mobile Broadband

June 30th, 2011by admin under M2M

New smartphones, tablets, and machine-to-machine devices are providing a compelling mobile experience by allowing people to engage their social networks, conduct business, and manage their day-to-day activities. This new, interactive mobile experience is bringing about a revolution in mobile broadband service models as service providers experiment with innovative ways to monetize mobile broadband, attract and retain customers, and efficiently manage network resources.

Join Tekelec for a webinar on July 13. The webinar will highlight four use cases that demonstrate how service providers can implement network and subscriber intelligence to monetize mobile broadband, including:
• Personalized service tiers
• Simplified services for the multi-device customer
• Adding value to over-the-top applications
• Casual usage and loyalty programs

To sign up, click here.

Policy Management Handbook

June 23rd, 2011by admin under Policy Control, Uncategorized

Tekelec recently released a reference guide that describes policy management solutions to enable cost effective deployment of broadband service, which guarantee Quality of Service (QoS) and generate revenue. Use cases include:
• Fair Use Management
• IMS VoLTE Call Control
• Location/Roaming

Download the reference guide here.

Yet More on Proxy versus B2BUA

June 22nd, 2011by Jiri Kuthan under SIP

It seems that for some readers I was too little technical on this subject in the past, so let me address that, particularly for transparency and feature richness.

First of all, what is it exactly lack of transparency, and why is it bad? In the Internet context, transparency means the network is guaranteed not to interfere with traffic between end devices. Transparency therefore affords innovation because new features will not be impaired by unexpected network behavior. More can be found in several RFCs, namely rfc4924 and rfc2775. Similarly in the SIP context, the network represented by SIP servers, is transparent if the servers interfere only little with traffic.

Behavior of a SIP proxy server is strictly governed by RFC3261: A compliant SIP proxy server only modifies few header-fields to mark the SIP message path (Via, Record-Route, Route) and that’s it. In the field, such a hard constraint turned out to be impractical particularly due to NAT traversal. In many cases, SIP traffic from behind NATs advertises un-routable addresses (affected message parts: SDP payload, Contact header field). Then “pragmatic” proxy server implementations choose to give up on strict compliance and change the un-routable addresses to routable ones.

The key difference in a B2BUA is that rewriting SIP traffic is not an exception, it is a rule. SIP proxy servers are bound to keep messages unaltered and implementations only divert from compliance when necessary. In contrast to that, a B2BUA is not bound to keep traffic unaltered by any standard. In fact, many B2BUA implementations are built upon the concept that altering as much as possible is a feature. Some apparently suspect this is sort of academic concern — well it is definitely not. Here is a very real-world implication of B2BUA: you simply cannot troubleshoot traffic. If a request is changed, while visiting a B2BUA, it is very hard to correlate incoming traffic to outgoing. Even if — in absence of unique call id — you try to correlate by phone numbers, there are numbers to which so many calls are routed that you will get lost in tons of traffic.

Similarly the argument that B2BUA produces features is fairly broken as it in fact tends to break them. A good example is the attempt to standardize reverse verification of incoming calls as a practice. The idea has been fairly simple: when a call is coming in and before your phone begins to ring, you ask the originating end devices if the call is genuine. The argument from B2BUA vendors has been you cannot do this because B2BUA’s way of mangling traffic will remove a valid reference between the original call and the verification request.

Shortly, B2BUA does aggressive changes to SIP traffic, and these changes have negative impact both on operational life and introduction of new features.

Some argue too that B2BUA creates new features. That may sometimes be the case, even though not always, see the reverse verification example. Less vaguely this is the case when nature of a feature requires an automaton to alter existing calls. The most frequent case I am aware of is network-terminated calls either on exhaustion of prepaid calling card credit or dead end-devices.

Keeping call information does not come for free though: it increases memory consumption and effort to replicate data to offer reliability.
This may sound too uninteresting in the age of cheap gigabytes of memory, however in high-density deployments that utilize servers’ memory to full extent it means buying more equipment, and being less “green”.

What I can therefore recommend is to use “pragmatic” proxy servers for better scalability and fewer side-effects unless you need to terminate calls from inside SIP-networks. It may be accommodating to find that some vendors offer servers that offer both “passive” proxy mode and active call-state approach, which gives you the possibility to change your operational mode if your network’s requirement change.

Key traits of an SDM system (Video)

June 16th, 2011by admin under Subscriber Data Management

Richard Bodin, product manager for SDM solutions at Tekelec, discusses the three characteristics of an SDM solution.

How is SDM being deployed? (Video)

June 15th, 2011by admin under Subscriber Data Management

Richard Bodin, product manager for SDM solutions at Tekelec, talks about the exciting ways that SDM solutions are being deployed in networks around the globe.

How is Subscriber Data Management Evolving? (Video)

June 14th, 2011by admin under Subscriber Data Management

Tekelec Product Manager Richard Bodin describes how SDM is evolving as networks themselves grow.

Wait a Second!

June 9th, 2011by Robert Sparks under SIP

SIP has several robustness mechanisms that leverage being able to say “Wait a bit before you try that again.”

A 486 Busy Here response can contain a Retry-After header field, allowing the endpoint to say “Please don’t try to call here again for 30 minutes,” based perhaps on knowledge it obtained from its user’s calendar.

A 500 Server Internal error can use Retry-After to say “Something’s keeping me from servicing this particular request right now, but please try again in 5 seconds.”

A 503 Service Unavailable error can use Retry-After to say something much stronger: “Something’s keeping me from servicing _any_ requests. Don’t send me anything more for at least 30 seconds.” As we’ll see in a moment, this is a very strong statement – one that needs to be carefully invoked.

The SIP Events architecture provides a way for an event server (such as a presence server) to say “I’m tearing down this subscription, and I need you to resubscribe, but don’t try to do so until at least 20 seconds have passed.” It does this with a NOTIFY containing a Subscription-State header similar to this:

    Subscription-State: terminated;reason=probation;retry-after=20

These mechanisms allow servers to avoid, and even redistribute, load. A registrar handling a burst of simultaneous registrations can quickly tell some or all of them to wait, using a different wait times to spread the returning load out a little.  One node in a cluster of presence servers can move its subscriptions to its peers by throwing all of its subscriptions into probation, as described above, again using a range of different wait times for different subscriptions. As the clients re-establish their subscriptions, the mechanisms for finding SIP servers can distribute the subscriptions among the peers. 

While the mechanisms are useful in the situations I’ve described so far, and may be exactly the tools an application relying on a limited external resource like a specialized DSP needs, they aren’t sufficient to handle the general case of overload protection. The granularity the tools work at is either very small (affecting this particular method applied to this particular resource), or very large (affecting all traffic between two elements). The IETF’s SOC working group is developing richer ways to help a server avoid being overloaded.

But even with those tools, there are situations where crushing load can appear before mechanisms at the SIP layer have a chance to help. Avalanche-restart scenarios, when whole campuses or even cities full of clients all come online at the same time due perhaps to restoration of power are a good example. In the extreme, action closer to the physical layer of the network (such as using firewalls to introduce the load in smaller increments) is warranted.

Finally, like most tools, using them without understanding what they do can lead to surprising results. Any code that generates a 503 Service Unavailable response, for example, deserves careful inspection. Some early proxy implementations make the mistake of forwarding 503 responses, when they should be taking a received 503 as input into generating their own final response. By blindly forwarding a 503, they are saying “Stop talking to me” instead of “I can’t find something that can handle this request,” which leads to unintended failures, such as the following:

Here, Alice and Bob are in a SIP dialog, perhaps for a phone call. Carol and Dave are in a separate dialog, either a different call or perhaps they have a subscription set up.  

503fig1

Something goes wrong with Bob’s UA and it has to return a 503 to a request it received.  Proxy 2 does the wrong thing and forwards the 503.

503fig2

Now Carol’s next request towards Dave can’t be forwarded through Proxy 2, even though there was nothing really preventing Proxy 2 from being able to service the request.  Carol and Dave have lost service unnecessarily and have no idea why.

503fig3

Alice (or anyone else whose requests towards Dave would have taken the path from Proxy 1 to Proxy 2) can’t reach Dave either.

Proxy 2 should have returned its own response, probably a 480 Temporarily unavailable, to the request that elicited the 503 from Bob’s UA. That way only the requests Alice was sending to Bob would be affected.

New whitepapers on LTE and data services

June 9th, 2011by admin under LTE, Mobile Data Pricing

The Path to LTE
This paper discusses how Service Providers can migrate their networks to all-IP LTE technology to help mitigate the effects of consumers’ never-ending appetite for bandwidth.

Monetization of Data Services
This paper highlights four use cases that illustrate how service providers can implement an intelligent approach to monetizing mobile broadband, including: personalized service tiers, simplified services for the multi-device customer, adding value to over-the-top applications, and casual usage and loyalty programs.

Vote for Best New Product of the Year

June 2nd, 2011by admin under Session Management

Tekelec’s Diameter Signaling Router (DSR) has been named a finalist for ‘Best New Product or Service of the Year’ by the 2011 American Business Awards (aka the Stevie Awards). Vote for the Tekelec DSR by visiting the People’s Choice Stevie Awards and entering short code T238L. Voting ends June 3.

The Tekelec DSR product centralizes routing, traffic management and load-balancing tasks to create an architecture that enables your IMS and LTE networks to grow incrementally to support increasing service and traffic demands. And, as the first point of contact at the network’s edge, the DSR is the ideal vantage point from which to defend your network against potential overloads or attacks. It can be deployed as an IETF Diameter Agent, 3GPP Diameter Routing Agent (DRA) or GSMA Diameter Edge Agent (DEA).

For more information on the DSR, explore the links below.

DSR Product Brief
Video: Why Choose a Tekelec DSR?
Video: Benefits of Diameter Signaling

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