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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
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Newport Networks

Newport Networks 1460 session border controller

SIP based SBC http://www.newport-networks.com/

Perhaps the First question to answer is: What is a Session Border Controller? This link will take you to a summary of the main functions performed by session border controllers.

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The Newport Networks 1460 session border controller is a new breed of IP networking equipment that combines the best of traditional circuit-switched services and the Internet to enable a VoIP provider to capture new revenue more quickly by delivering increased value to their subscribers. Session Border Controllers, are an essential component of the infrastructure required by Service Providers to deliver successful public voice and multimedia services over IP.

NAT traversal for SIP

NAT traversal in a SIP environment is the ability for a SIP call to successfully traverse a NAT (Network Address Translation) device at the border between two networks. Traversal of NAT devices is a problem that SIP, and other VoIP protocols, must overcome if they are to be deployed in a public network. NAT devices perform network address translation between private and public IP networks. SIP clients residing in private networks use private IP addresses in SIP messages which results in the far end SIP signalling and media to be unroutable. Session controllers are designed to overcome NAT traversal issues, amongst other things. Deployed within the carriers network the 1460's Automatic Channel Mapping feature (ACM) allows SIP services to be delivered to large numbers of customer sites, even if there are multiple NAT devices in the access and customer networks. Read a White Paper on solving NAT traversal problems here.

White Papers

Newport Networks has written a number of White Papers on the subject of session controllers, including: NAT traversal, SIP security, and SIP peering. These are freely available on line at www.newport-networks.com/whitepapers and can be viewed on line or downloaded in PDF format.

SIP Security

With all the press activity recently regarding VoIP fraud the security of SIP services has never been so important. We have put together some White Papers that examine different security aspects:
  • SIP Security Start here if you want an overview of the issues.
  • SIP Security and IMS White paper that looks at the issues that Service Providers face in keeping their core network up and running in adverse conditions.
  • IPsec and VoIP Networks White Paper that looks at IPsec and why it won't traverse a NAT. Examines TISPANs selection of UDP encapsulated IPsec to overcome these problems.

Regulatory requirements

The FCC and the EU are mandating VoIP service providers support Emergency Call Handling and Lawful Interception. These papers look at some of the options in addressing these needs:

VoIP Bandwidth Calculation

Calculating how much bandwidth a Voice over IP call occupies can feel a bit like trying to answer the question; "How elastic is a piece of string?" However, armed with a basic understanding of the parts that make up the whole, the question becomes easier to understand. This White Paper examines the process that turns voice into Voice over IP. Also see the on-line VoIP Bandwidth Calculator.

Newport Networks has offices in Caldicot and High Wycombe the UK, and sales and support offices in France, Germany, China, USA and Canada.

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Created by jht2, Last modification by daveg1k on Tue 13 of Jun, 2006 [10:08 UTC]

Comments Filter

VoIP Security Solutions

by jenniferhan on Thursday 27 of December, 2007 [07:02:50 UTC]
SpeedVoIP is a professional VoIP Security and VoIP anti blocking solutions provider.
The core solution for VoIP Security and VoIP anti-blocking is VGCP (VoiceGuard Control Protocol).
It can work with any 3rd-party Softphone / ATA / Gateway / IP Phone / IADs and SIP proxy or server.
It can work in the way similar to that of SOHO router, but it only encrypts and decrypts SIP and RTP packets on link layer, not to handup these packets to IP stack for forwarding while bypassing other data packets originating from SIP terminals. In this scenario, peak throughput and minimal CPU overhead can be easily achieved.

VoiceGuard can real-time incorporate light-weight traffic for puzzling and bypassing VoIP blocking system without consuming more bandwidth and compromising voice quality. Even in some circumstance, VoiceGuard can simulate traffic behavior of universal data networking protocol such as OICQ, MSN and so on.

For more information, please refer to: http://www.speed-voip.com/index-36.html

Andy
xd.wong@speed-voip.com
andywong-01@hotmail.com

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