All posts under tagged ‘TransNexus’

Feed for all posts filed under "TransNexus"

OpenSER VS SER, running on a $3,000 Lintel server, both manage 8 billion minutes of VoIP traffic per year.

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com via eMediawire.
Two competing open source projects have now been compared in a side by side test. SIP Express Router, also known as SER (www.iptel.org), is the respected pioneer of open source SIP proxies. SER has been available since November 2003 and has a reputation for high performance and reliability. The upstart, OpenSER (www.openser.org) was created when developers disappointed by SER’s slow release schedule, forked a version of SER to create OpenSER in June 2005.
Ever since, SIP proxy users have been faced with the question, which project is best?

TransNexus, an independent developer of VoIP Operations and Billing Support System (OSS/BSS) software decided to answer this question for its customers once and for all. Most product benchmark test plans are designed to yield incredible results for marketing promotion. The TransNexus benchmark plan was different. It was designed to mimic a production network with a lot of failed call attempts and the added overhead of call detail record collection.

As expected, the TransNexus results were not as amazing as some of the other published test results, but they were realistic and still very impressive. Telecom equipment vendors need to beware, both SER and OpenSER are ready to scale for the largest wholesale carrier operations. To summarize the results, TransNexus found that either OpenSER or SER SIP proxies have the performance to manage 720 calls per second on a $3,000 Linux server with dual Intel Xeon CPUs. For a typical wholesale carrier operation, this performance is sufficient to manage over 700 million minutes of traffic each month! While OpenSER and SER are competing against each other, they are rapidly out running the cost/benefit performance offered by commercial telecom equipment vendors.

Here is What Transnexus had to say;

Performance Results for OpenSER and SIP Express Router

We often hear the questions:

  • How fast are OpenSER or SER in a real world environment?

  • How do SER and OpenSER compare?

We decided to answer these questions and created a detailed performance test for OpenSER and SIP Express Router. To simulate a production environment, the SIP proxy under test queries an external OSP server for routing information on each and call and then reports call detail records to an external OSP server. Five destinations are returned to the SIP proxy for each call in random order. Four of the five destinations simulate call failure scenarios so the SIP proxy must retry the call an average of two times before the call is completed. These tests were performed on a single core of an Intel Xeon 5140 2.33 GHz CPU.

Here is a brief summary of what transnexus learned. For all the test details, click here.

  • The performance of OpenSER V1.2 and SER 2.0 are not materially different, however, there are two minor differences.
    • SER V2.0 requires less memory.
    • OpenSER V1.2 has less post dial delay.
  • By all measures, OpenSER V1.2 is significantly better than OpenSER V1.1.

  • For production operation (with call retries and CDR reporting), we suggest the following simple guideline for sizing server hardware to operate at 60% CPU utilization for OpenSER V1.2 and SER V2.0:

One GHz of CPU processing capacity can manage 60 calls per second.

For example, a server with two, dual core, 3.0 GHz CPUs would effectively have (2 CPUs * 2 cores * 3 GHz per CPU) twelve GHz of CPU processing capacity. This server, hosting either OpenSER V1.2 or SER 2.0, would be able to manage 720 calls per second at approximately 60% CPU utilization.

CPU Utilization

The following chart plots CPU utilization as a function of calls per second. The results for OpenSER V1.2 and SER 2.0 are identical. The performance of OpenSER V1.2 is 13% better than OpenSER V1.1.

image002.gif

Memory Usage

Memory is not a major resource requirement, even at high loads. SER 2.0 has the lowest memory requirement.

image004.gif

Post Dial Delay

The data on the following chart is an indirect indication of Post Dial Delay (PDD). The data presented is the percentage of calls in each test that experienced a PDD greater than six seconds.

image006.gif

Call Completion

The following chart presents the percentage of calls which were not completed successfully for each test. When CPU utilization was less 90%, both OpenSER V1.2 and SER 2.0 completed all calls successfully.

image008.gif

Published on May 24th, 2007 under , , ,

Secure multi-lateral VoIP peering software published to Sourceforge as Open Source

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com

Atlanta, Georgia (USA) –– Feb 14, 2007. TransNexus, Inc. has made the OSP Toolkit and RAMS open source projects publicly available on SourceForge. The OSP Toolkit is a client side implementation of the OSP peering protocol. The OSP Toolkit, written in C, is a mature open source project begun in 1999 and has been integrated into numerous commercial and open source VoIP products. The RAMS OSP server is a java based OSP server developed for testing and as a reference implementation.

The Open Settlement Protocol (OSP) is an IP Operations and Billing Support Systems (OSS/BSS) protocol defined by the European Telecommunications Standards Institute (ETSI TISPAN), www.etsi.org. OSP is officially known as ETSI Technical Specification 101321 for inter-domain pricing, authorization and usage exchange. OSP is unique because the way it uses PKI services to enable secure peer to peer communication between VoIP networks. “The OSP protocol was developed to enable direct multi-lateral peering among VoIP networks. OSP provides secure inter-domain access control and eliminates costly network bottlenecks”, stated Richard Brennan, Chairman of ETSI TISPAN Next Generation Network Architecture working group.

RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute’s OSP Peering protocol (ETSI TS 101 321).

The OSP Toolkit and the RAMS OSP test server is available at links provided below. A free version of the TransNexus commercial OSP based peering server is available at www.transnexus.com.

Links;
The OSP Toolkit
RAMS server
www.transnexus.com

Published on February 15th, 2007 under , , , , , ,

Open Source Software for VoIP Routing

Source: voipcentral.org


TransNexus, Inc. has brought forth a new routing software that happens to be open source code one.
The software happens to be Gatekeeper Transaction Message Protocol (GKTMP) module which comes under the category of free BSD (Berkeley Software Distribution) license
The attractive features in this software is that it provides an open interface to the TransNexus commercial route server with profit optimization features such as least cost routing, quality of service routing, automated credit controls and secure access control features based on digital certificates and public key infrastructure (PKI) services.
The new GKTMP module is more versatile as it is written in C. The testing has been carried out extensively on Linux and Sun Solaris operating systems.
There is functionality in it that is it can be used with open source peering servers for creation of reliable and feature rich peering server for Cisco VoIP networks.

For more read


Member of "Hype Media! Network"