String around my finger
A 'blog to remember things I found.
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FWD# 61159
Wed, 03 May 2006
Voipbuster - the Free Calls Company
Warning: These Web pages use a nasty Cross Browser marquee from Dynamic Drive that sucks all your PC's CPU time !!! Don't leave this site ticking away in a browser tab or two !!!
Offers Free calls to landlines in many countries including Australia, and free VoIP-In numbers in e.g. United Kingdom. You need to install and run the Windows client to create yourself an account.
" You can now make FREE calls to several countries. These calls are limited to 1 minute per call. If you want UNLIMITED calls to our FREE destinations please go to our website and buy 10 Euro worth of credit. "
* Max 20hrs per month of free calls.
Rates to other countries are e.g. 1 (Euro cent?) per minute to USA.
SIP settings:Username: Your VoipBuster username Password: Your VoipBuster password SIP/Proxy registrar: sip.voipbuster.com Domain/Realm (optional): voipbuster.com STUN server: stun.voipbuster.com
Always dial 00 + countrycode + areacode + subscribernumber.
?? Test numbers
I've re-subscribed to oztralia.com for inbound Fax-to-email (I've cancelled my 2nd Optus landline) and for testing Asterisk trunking.
The web site hasn't improved much -- all the detailed information is still in the forums.
Their plans have been simplified down to just two, plus added services.
Whirlpool - Australin Broadband News links
This post is to collect up VoIP links on Whirlpool.
Forums - Internode
Discussions about Nodephone, Billion 7402VGP and
Sipura
gateways ...
Wiki
Links to FAQs, including
Wiki - VoIP Providers
and
Wiki
- VoIP
Forums - Voice over IP
Includes
General VoIP Information and Reference,
How to unlock engin VoIP Box (Sipura SPA2000),
Spa 3k dial plan (& 2k, linksys pap2 &c),
VOIP Provider Wiki,
Australian Asterisk VoIP Users Group,
Forums - Voice over IP - Asterisk
.
I installed V2.7 on an old IBM Personal Computer 300PL/. Asterisk@Home is an interesting IP PBX based on CentOS and chock-full of VoIP goodies.
It does take a little while to get to grips with everything that is included, and work out how to drive it all.
It is a IP PBX with SIP and IAX interfaces, rather than a SIP router. It wants to relay all the voice traffic as well as the signalling. This probably has a performance overhead, but does make gatewaying easier.
I think the best setup guide is the "Asterisk@Home Without Tears - The Dumb-Me Guide" which is discussed at Whirlpool Forums - Voice over IP - Asterisk - Kickstart for Asterisk newbie.
Mon, 22 Aug 2005Files available via FTP. Latest README is for version 0.9.3, however, I'm using last-available Red Hat RPMs for version 0.8.12.
AARNet Tutorial on SIP and SIP SER server has links to useful configs.
Installed ser-0.8.12-0.i386.rpm, ser-mysql-0.8.12-0.i386.rpm, mysqlclient10-3.23.58-6.i386.rpm, along with php-5.0.4-10.3.i386.rpm, php-gd-5.0.4-10.3.i386.rpm, php-mbstring-5.0.4-10.3.i386.rpm, php-mysql-5.0.4-10.3.i386.rpm, php-pear-5.0.4-10.3.i386.rpm and phpMyAdmin to manage MySQL.
I read Getting Started with MySQL, 2.9.2. Unix Post-Installation Procedures, and 2.9.3. Securing the Initial MySQL Accounts.
Opinions about MySQL root password differ -- see Step by Step Installation of SER on RedHad Fedora Core 1 v. phpMyAdmin error messages, and SIP.edu Cookbook - Configuring SER recommends using the PW environment variable to avoid being prompted all the time.
Set domain in dhcpd.conf, set line1_* and line2_* in SIPmacaddress.cnf, and added accounts using serctl. Phone now registers to SER.
Aug 22 10:12:14 tower ser: DEBUG SIP: Incoming SIP Method=REGISTER cseq=101 request uri=<10.1.1.50> src ip=<10.1.1.222> from uri=<sip:1002@10.1.1.50> to uri=<sip:1002@10.1.1.50> from tag <<null>> to tag <<null>> header <<sip:1002@10.1.1.222:5060>> Aug 22 10:12:14 tower ser: DEBUG SIP: Incoming SIP Method=REGISTER cseq=101 request uri=<10.1.1.50> src ip=<10.1.1.222> from uri=<sip:phone2@10.1.1.50> to uri=<sip:phone2@10.1.1.50> from tag <<null>> to tag <<null>> header <<sip:phone2@10.1.1.222:5060>> Aug 22 10:12:14 tower ser: DEBUG SIP: Incoming SIP Method=REGISTER cseq=102 request uri=<10.1.1.50> src ip=<10.1.1.222> from uri=<sip:1002@10.1.1.50> to uri=<sip:1002@10.1.1.50> from tag <<null>> to tag <<null>> header <<sip:1002@10.1.1.222:5060>> Aug 22 10:12:14 tower ser: DEBUG SIP: Incoming SIP Method=REGISTER cseq=102 request uri=<10.1.1.50> src ip=<10.1.1.222> from uri=<sip:phone2@10.1.1.50> to uri=<sip:phone2@10.1.1.50> from tag <<null>> to tag <<null>> header <<sip:phone2@10.1.1.222:5060>>
Original App Load ID: P00303010411 Boot Load ID: Version: 3.1(4.11)
Very old SCCP image -- current is 7.x
Tries to fetch: OS79XX.TXT SEP003094C2B25D.cnf.xml XMLDefault.cnf.xml
Converting a Cisco 7940/7960 CallManager Phone to a SIP Phone and the Reverse Process
Cisco 7940 and 7960 IP Phones Firmware Upgrade Matrix
Upgraded to SIP ver 6.3 by putting P0S3-06-3-00 in OS79XX.TXT and "image_version: P0S3-06-3-00" in SIPDefault.cnf.
Maybe try version 7.5 later ...
Cisco IP Phone 7960/7940 User Guide for SIP
Cisco SIP IP Administrator Guide, Version 7.5
SIP IP Telephone 7940/7960 Software
See also Appendix D - Configurable Parameters for the SIP IP Phone (Versions 6.x and 7.x) for definition of fields in SIPDefault.cnf -- linked to by AARNet tutorial Configs for Cisco 7960 with SER.
SIP - Session Initiation Protocol
Cisco IP Phone Administrator Guides for SIP
Mon, 15 Aug 2005
FWD 1800 gateway in OZ available
|
"sandman" wrote:
We've made available a **FREE** 1800 number FWD gateway to see if people would find it usefull. A few technical matters to make note of 1) No warranties implied, its experimental 2) The gateway is registered as an account on FWD 3) Your CLID will show up at the remote end 4) We do not know what call depth is possible at the FWD end on a single account 5) If your calling an OZ FWD number, if that person is registered as an IAX user, there should be **NO** latency 6) PREFIX to use infront of FWD number is 393 ie: 393xxxxxx where xxxxxx is the FWD number u need to call. 7) Call 1800262218 Enjoy please and provide feedback. I have no idea how its going to go and this is experimental. If its well received, we may make it available long term... CIao
It works!! I phoned my laptop from my work phone. SJphone console says:
Incoming Call Incoming call from 0399054774 from 69.90.155.70 : 5060 SIP session using Asterisk
65.39.205.121 and 69.90.155.70 both belong to Peer1 Internet Bandwidth Inc. and are reached through NewYork.savvis.net.
http://forums.whirlpool.net.au/index.cfm?a=wiki&tag=voip_1800gateway http://account.freeworlddialup.com/index_new.php?section_id=97 http://www.freeworlddialup.com/content/view/full/333/
325 faktortel 335 fire 373 free 393 fwd
Sun, 14 Aug 2005
|
Our flagship product is SJphone which is the best standard-based softphone for MS Windows, PocketPC, Linux, Mac OS X, web-applications etc.
It supports both SIP and H.323 standards, many features and comes with Internet Telephony Service Providers (ITSP) profiles, e.g. Linux/MAC OS Free World Dialup Profile, and a guide for creating your own profiles.
I found this application to be quite techie-friendly with "Info" pane, Options -> Support -> Bug Reporting -> Start Recording / Submit, forums where developer replies etc etc. Not a pretty application, though there is some mention of "skins".
I "Registered", but I'm not sure if that enables any more functionality. Provided CODECs: GSM 6.10, iLBC 30ms and 20ms, Speex 15.2k and 15.2k 40ms and 8.0k 40ms, G.711 A-Law and u-Law. The purchased Windows SJphone Plus version comes with G.729.
Codec info page and Voice Over IP - Per Call Bandwidth Consumption.
Wed, 03 Aug 2005US$399.99 Broadband Internet video phone using high speed Internet connection.
They sell something that looks like a Sipura SPA-3000, which contacts their server to see whether it can route calls over VoIP, or falls back to using the PSTN.
It all starts off looking great:
1. PhoneGnome is a "thing". It is a revolutionary concept. FREE calling is a feature built into a piece of hardware that you purchase and own. There is no signup, no "account", and no monthly fees. ...
4. Give your current phone service a kick with advanced capabilities you can't even get from the phone company, like voicemail-to-email, online phone book with click-to-dial, and more, all included FREE with your one-time purchase of PhoneGnome ...
7. No obligations, No risk, full-moneyback guarantee.
However, when you get down to the PhoneGnome License Terms and Conditions of Use Agreement
Use of the PhoneGnome Services ...
(d) You may not reverse engineer, distribute, publish, display, modify or in any way exploit the configuration parameters which we provide as a means to access the PhoneGnome Services.
(e) You may not use the PhoneGnome Services for telemarketing, broadcast fax, or to deliver unauthorized or unsolicited advertising, promotional materials or solicitations.
(f) You acknowledge that the PhoneGnome and any embedded software or firmware is exclusively for use with the PhoneGnome Services.
(g) You agree not to perform a factory reset of the PhoneGnome hardware.
(h) You may not use the PhoneGnome Services with or connect the hardware component of the PhoneGnome Servcie to a PBX or automated dialer.
It doesn't look like a "thing" that I could configure to use with a home Asterix PBX setup for example.
Also check out the Privacy Policy where they get to record and analyse your PSTN as well as VoIP call information.
Wed, 27 Jul 2005Computer Telephony hardware vendor with Open Source drivers. Located in Adelaide / Sydney.
Products include OpenLine4 (4-port FXO), OpenLog4 (4-port logging), OpenSwitch6/12 (FXO or FXS), and OpenPRI (E1).
Their Open Source softwaree includes drivers, OpenPBX, OpenH323 PSTNGW, CT (computer telephony) Server, Logger, and Unified Messaging Server.
Once past the survey you can download interesting-looking things.
Unfortunately, their hardware seems too expensive for home experimentation, e.g. OpenSwitch6 885 Euro, or OpenLine4 550 Euro.
Sun, 24 Jul 2005Mark A. Miller wrote a series of articles, published on www.voipplanet.com
- Part I—History and Architecture
- Part II—Protocol Capabilities
- Part III—Message Formats
- Part IV—The Session Description Protocol (SDP)
- Part V—SIP Signaling
- Part VI—Testing SIP Interoperability
Each article is brief and easy to read, but gives a good overview, and pointers for where to look for more details.
More backgrounder articles, archives.
Fri, 24 Jun 2005Xten make SIP Softphones. X-Pro was their commercial product, and X-Lite was their free version (without conferencing, forwarding etc). X-Pro seems to be still available from resellers. Xten's new product is now eyeBeam Voice, Video & IM Softphone which seems to be available free as a walled-garden ineen.
Of all these, only X-Lite is available for Linux. I downloaded "X-Lite v2.0 Build 1105d for Linux" X-Lite_Install.tar.gz. Inside is just "xtensoftphone" and "README". You may need to be careful that you have all the libraries that xtensoftphone requires - no dependency checking like installing rpm's.
Running "./xtensoftphone" checks network connection, runs audio wizard, and then takes you to Menu / System Settings / Sip Proxy / Default to enter your details. See X-PRO Quick Start Guide. Full details of what to enter are in the X-PRO User's Guide section "3.2.2 Logging in to Your Provider" on page 8 - "Authorization User" and "Out Bound Proxy" may not be needed.
I'm using a free SIP account from SIPphone.
Next task is the Phonebook. Either enter it using the menu, or load from a CSV file
Proxy ID,Name,SIP URL,Phone Type,Speed Dial 0,"Sipphone Audio test number","1-747-474-3246",0,"" 0,"Sipphone Hear your number repeated back to you","**",0,"" 0,"Sipphone Welcome message","1-747-474-5000",0,"" 0,"Sipphone Diagnose router","*0",0,"" 0,"Sipphone 1-800-555-TELL","411",0,"" 0,"Sipphone Free Home Tel","05161398491150",1,"" 0,"Sipphone Free Home Fax","05161398491151",1,""
Anyway, it all works. I was able to make free 1-minute calls to landlines in Australia.
NAT STUN traversal http://ip.xten.net/
Wed, 18 May 2005
Shopping for the Best VoIP Service
Advice for US residents, but some of the concepts are probably relevant to Australia.
See links to other pages too.
Mon, 16 May 2005DrayTek make ADSL/Cable VoIP routers, and Internet Telephony Adapaters.
warcom.com.au has many DrayTek products including Vigor 2100V (broadband router, telephone and POST ports) for A$189.00 +$9.00 delivery, and Vigor 2900VG (broadband router, 4-port switch, 802.11g, USB print server, twin FXS ports) for A$349.00 + $9.00 delivery, and VigorTalk (single FXS voice adapter) for A$149.00 + $9.00 delivery.
Linksys make voice products including RT31P2 Broadband Router with 2 Phone Ports and WRT54GP2A-AT Wireless-G Broadband Router with 2 Phone Ports and PAP2 Phone Adapter with 2 Ports for Voice-over-IP
PAP2 available index at shopbot.com.au
Avaliable at OzTechnologies (other Billion, Draytek and Engin products too).
At warcom.com.au PAP2 is A$99.00 + $9.00 Delivery, and WRT54GP2 (Wireless-G Broadband Router, 3-port switch and 2 phone ports) is A$180.00 + $9.00 delivery.
Sipura available index at shopbot.com.au
Sipura SPA-3000 (with PSTN gateway) available at Voxilla Store for US$96.95 + $3.95 + UPS World Wide Express ($65.70). Ouch!
warcom.com.au have several Sipura models for sale
SPA 3000 from warcom.com.au for A$185.00 + $9.00 shipping.
Fri, 29 Apr 2005
NodePhone - Voice over Broadband
Internode plan to launch NodePhone services during May 2005.
From the press release
Internode will recommend a range of standards-compliant telephone attachment hardware tested with the service and supported by Internode. This will include an Integrated Access Device (IAD) option, eliminating the traditional "rats nest" of boxes, cables and power-packs. The IAD does this by placing two simultaneously available voice lines, an ADSL1/ADSL2/ADSL2+ firewall-equipped router, a 54Mbps wireless access point and a four-port 100-megabit switch in a single compact box. Another supported option will be a single line phone attachment device that is plugged into an existing broadband connection.
This is the first mention I know of for a router plus VoIP gateway box (plus VoIP service) being sold by an ISP in Australia.
Who Sets the Standards for VoIP?
From the article:
In summary, an understanding of the underlying standards should help network managers sort through the various systems and products that they are considering for VoIP deployment on their network. Products that adhere to ITU-T standards, such as H.323, are most likely to have originated from a telephony and circuit switching perspective. Conversely, products that adhere to IETF standards, such as SIP, are most likely to have originated from the data and packet switching side of the house. Both are quite workable, but approach technical issues such as connection setup/disconnect in different ways. Adopting an architecture that leans in one standards direction or the other, however, can help focus all product decisions down the same road, and thus bypass some of the interoperability challenges that you would prefer to read about, rather than experience first hand.
So it's like choosing Microsoft file and print or Netware file and print. Both methods provide similar functionality to the users, but are different underneath to deploy and manage. Both have different vendor lock-in, etc. Choose one or the other rather than both simultaneously plus glue.
PS. Avoid choosing some non-standard that looks similar, but is doomed to fade away.
Sipura make ATAs and IP phones.
Sipura SPA-2100 dual analog ATA and home router.
Sipura SPA-3000 (with PSTN gateway). Dual analog ports can be used in FXS or FXO / VoIP Line Sharing mode.
Friday, 08 April 2005 - Sipura SPA-841 Phone Review
Thursday, 14 April 2005 - Sipura SPA-1001 ATA Review
April 26, 2005 - Cisco announce purchase of Sipura for its Linksys division.
Thu, 28 Apr 2005sipsak is a small comand line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can make several different tests, send the contents of a file, and interpret and react on the responses. It supports (de-) registration with given contact URIs and digest authentication.
Thu, 31 Mar 2005Windows messenger claims to enable "video conversations" and to support SIP.
However, first some terminology. Windows Messenger is also known as .NET Messenger. However, MSN Messenger is not the same thing at all. Many people like the extra eye candy in MSN Messenger. Go to Help / About to find out exactly what you have. You may be able to run both, but not at the same time.
Getting Windows Messenger. http://www.microsoft.com/windows/messenger/ is subtitled "Windows Messenger Update for Windows XP". "Note: The latest version of Windows Messenger is available for Windows XP in Windows XP Service Pack 2." (This may be Windows Messenger 5.0).
There is also Windows Messenger for Microsoft Office Live Communications Server 2003 and Microsoft Office Communicator 2005 and Exchange Instant Messaging (IM) Client 4.6 (for Windows 95, Windows 98, or Windows ME).
Inside Windows Messenger - How it Communicates This article discusses ... Session Initiation Protocol- (SIP) based server options.
The latest seems to be Windows Messenger 5.1, dated 3/24/2005.
Wed, 30 Mar 2005Features: ... console version, SIP, DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses), includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.
Now version 1.0.0, uses eXosip.
Siproxd - a masquerading SIP proxy
Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT).
It allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
Includes a two-way RTP proxy!!! May run on WRT54G.
Read the manual for lots of configuration examples, including running "in front" of a NAT router, and transparent proxying.
May also be suitable for Monash, where only some machines have direct Internet access, but there isn't any NAT.
oSIP is now at The GNU oSIP library. SIP links including eXosip, Partysip, antisip.
Tue, 29 Mar 2005I have previously used GnomeMeeting in H.323 mode.
New version will support SIP too - upgraded to version 1.2.1 -- but it might require version 2.0 or a CVS build.
From home, NAT detection (using 83.103.82.86 / stun.voxgratia.org) reported
STUN test result: Port Restricted NAT.
Using a STUN server is most probably the most appropriate method if your router does not natively support H.323. Notice that STUN support is not sufficient if you want to contact H.323 clients that do not support H.245 tunneling like Netmeeting. In that case you will have to use the classical IP translation and port forwarding.
Worth reading: Inside GnomeMeeting interview.
http://mail.gnome.org/archives/gnomemeeting-devel-list/
Check http://mail.gnome.org/archives/gnomemeeting-devel-list/2005-March/msg00134.html http://fedora.roving-it.com/gnomemeeting-1.3/ yum-able ??
http://wiki.seconix.com/doku.php
Sat, 26 Mar 2005Welcome to Vovida.org - a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments.
Applications: VOCAL B2BUA Load Balancer MPEG4IP Open G.729(A) Open AMR OpenOSP RGL SIPRG SIPSet SIPTiger STUN Server WinRTP
Protocols: SIP COPS GDOI MGCP RTSP RTP RADIUS SRTP TRIP
Website not updated since 16 March 2004.
Fri, 25 Mar 2005Spring 2005 VON Conference & Expo
"A team of Google honchos met this week with several Net telephone service providers, sources familiar with the talks told CNET News.com, renewing speculation that the search giant may be exploring a move into the fast-growing market."
Voice on the Net Australia August 15-17, 2005.
Build Your Own PBX using Asterisk@Home and Digium Wildcard X100P FXO card from eBay.
Configuring Asterisk@Home For BroadVoice
Thu, 24 Mar 2005Sells products that include "media exchange", SIP gateways, switches, hard and soft IP phones.
Office in North Sydney
Zultys Launches New Family Of IP Phones
LIPZ4 Linux SIP Softphone datasheet. Download softphone-1.3.13-0.i386.rpm after fill in form
Some problems:$ ./KylixPhone ./KylixPhone: symbol lookup error: ./KylixPhone: undefined symbol: initPAnsiStringsMight be the Kylix run-time environment: forum: Installation problems with Fedora Core 1 and Core 2, Redhat Linux 9.0 and softphone, fedora. Wed, 23 Mar 2005
Minisip is a SIP User Agent ("Internet telephone") developed at KTH currently running on Linux. Keywords: Secure VoIP; SIP; MIKEY; RTP; SRTP; SDP; Video Telephony; Push-to-talk.
http://www.gnomemeeting.org/index.php?rub=3&pos=0&faqpage=x213.html#AEN294 Getting it to work behind a firewall
http://www.wifi.com.ar/english/voip.html Linux LiveCD VoIP Server $349
http://www.iptel.org/ser/download/ Download iptel.org SIP Express Router
http://www.netcraft.com.au/products/converse.php VoIP Application Suite
http://hans.fugal.net:2500/linuxaudio/show/Network+Audio Network Audio catalogue
http://www.pbs.org/cringely/pulpit/pulpit20050303.html The Best days of Voice-over-IP Telephone Service May Already Have Passed -- telcos preferrentially tagging their own VoIP packets.
http://theswitchboard.ca The Switchboard, a VoIP applet Cringley finds very clever. Not sure if it is SIP.
https://jain-sip-applet-phone.dev.java.net/ A JAIN-SIP Applet Phone for the People! This SIP user-agent with presence (coming soon), instant messaging and audio support (either real time with RTP or voice messaging using TCP). If your firewall will let you do it, you can use udp and do real-time voice. Else, you can tunnel voice, IM and signaling through TCP. This application can be launched as stand alone user agent or embedded in an applet (the web application to access the applet is provided along with a customized version of the gateway). Requires installation of the java media framework.
Tue, 22 Mar 2005Start at: Voice and IP Communications
Cisco SIP Proxy Server and Cisco SIP Proxy Server Data Sheet (PDF)
SIP (Session Initiation Protocol) contains links to lots of white pages.
High-Availability Solutions for SIP Enabled VoIP Networks
Cisco IOS SIP Configuration Guide
IOS 12.2 SIP Gateway Enhancements
There's another thing called Cisco CallManager Express which is a call-processing application in Cisco IOS software. Previously known as Cisco IOS Telephony Services (Cisco ITS).
Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2
Cisco IOS Voice Configuration Library, Release 12.3
Fri, 11 Mar 2005http://www.oztralia.com/ Mesh of Web sites - Net2MAX / Oztell / oztralia that provide VoIP services.
Choose a package, and add extra modules if needed. Requires a $20 deposit to get account enabled, even for $0 per month "Iron Package".
A "Bronze Package" ($2.95 per month) plus "Fax - Fax To Email" ($2.95 per month) module will give you a basic indial FAX to email-of-account-holder service.
To do more-complicated routing requires a "Storage - Multimedia Email" module which has filter rules (extra $1.95 per month). This is now close to the cost of a Silver Package which gives an extra dialin number.
Various HOWTOs are hiding in the Ozforums.
Thu, 10 Mar 2005Big list of SIP, H.323 and MGCP IP-Telephony Products
OzVoIP Australian residential and SOHO voice over IP specialist.
ATP Distributor for Digium, Asterix, VoIP products, training, VoIP termination, ... located in Glen Waverley.
Thu, 24 Feb 2005Voxilla News, 'How-Tos', product reports and reviews, and communtiy forums.
VOIP-info.org The VOIP Wiki - a reference guide to all things VOIP.
iptel.org iptel.org is a VoIP portal promoting VoIP technologies and is initiated by Germany's national research institute Fraunhofer Fokus.
Thu, 02 Dec 2004http://www.aarnet.edu.au/events/conferences/2004/sip/
http://noc.grangenet.net/documents/workshops/information?id=206
http://www.iptel.org/ser/
http://www.voip-info.org/wiki-SIP+Express+Router
Sun, 21 Nov 2004http://testyourvoip.com/
TestYourVoIP.com will make a call from wherever you are to one of our
U.S. or international test locations and report the results for free.
It'll only take about 20 seconds if you have Java installed.
Does MOS Analysis to and from one of several call destinations. Round-Trip Latency / Packet Discards / Packet Loss / Loss Periods / Jitter.
Also Signalling delays.
Fri, 08 Oct 2004http://www.research.earthlink.net/p2p/
EarthLink SIPshare, a simple, SIP-based proof-of-concept content sharing
application, demonstrates the viability of SIP as a protocol over which
peer-to-peer (P2P) applications other than the well-known voice and
video cases may be implemented.
SIP is like OSI H.* protocols, only done using IETF http-like syntax, proxies, LDAP directories etc etc.