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Michael Robertson Unveils SIPphone
JimCricket writes "After almost a year of preparation, the person behind MP3.com and Lindows has unveiled his latest venture: SIPphone. According to a CNET article, the new company sells VoIP-based telephones. I wonder what kind of latency you get with these devices." Interestingly, the CNET article reveals the telephones "...can only call other phones that use the same technology."
Everyone with broadband has a PC witha sound card anyway, so may as well just use Teamspeak.
The unofficial
From their web page, it looks as if you can call any other SIP phone:
Finally, you will want to call and talk to a live person. There are already thousands of numbers you can call. If you know a SIP number of a friend, you can dial it now or look up numbers on the SIPphone white pages. SIPphone lets you call users on other popular SIP services like FWD and Iptel as well. All SIPphone numbers will begin with 747 (it's optional to dial the first three numbers).
From the FAQ:
Q: Can I use software or what is called a softphone to make and receive calls with SIPphone?
A: Although it may work, at this time we cannot offer support for anything but a certified SIP phone.
Q: Are there other SIP phones I can order besides those offered at SIPphone?
A: The SIP phones offered at SIPphone are designed to work out of the box with SIPphone with zero or minimal configuration. We also work to offer the most affordable SIP phones available in the world. Many SIP phones cost hundreds of dollars. SIPphone sells 2 phones for just $129.99. It may be possible to use the SIPphone directory with other phones, but no technical support is available at this time to support this.
Q: I already own a SIP phone and I would like to use your SIPphone directory service. What should I do?
A: First, you need to sign up with our service at SIPphone Sign Up. These are the settings that you will want to use:
SIP Server: proxy01.sipphone.com (130.94.123.252)
STUN Server: stun01.sipphone.com (69.0.208.27)
NTP Server: ntp01.sipphone.com
TFTP Server: tftp01.sipphone.com (130.94.123.253)
Currently the SIPphone directory service has only been tested with the Grandstream BudgeTone 100 phone. Please check back for further updates on "SIPphone friendly" SIP phones.
Some things are more important than an animated rat
SIP is an IETF standard for voip, surely he just meant it could only phone other SIP phones! no need for any conspiracy theories! SIP is an open standard, and you can even get linux software linphone to use it... Just need a gateway to the traditional phone system and yer sorted.
"I wonder what kind of latency you get with these.."
I would assume the same latency you would have with any application that would have taken the same network path as the 'net phone's packets?
Perhaps you are talking about an audio delay? In that case, assuming your ISP has proper routing, there should be no significant delay (around the same as many cell phones) when speaking to someone else in your same country.
I've set up vbrick devices to use two T1's bridged for LAAtlanta conferences and the delay was barely noticable.
Not since dialup on a 28.8k modem have I noticed much problem with audio communications on the web. Definitely better than the telco's international service back in the 80's. I remember talking to friends in Germany and Japan and having to stop for long periods of time between sentences to prevent cross talk.
I think this product is so-so, though. Without a subscription based access from the voip phone to a telco bridge and a real phone number, it's not going to explode in popularity regardless of it's audio quality.
SIP is not limited to just VoIP, as the name says it is Session Initiation Protocol. There already is a reasonable GNU SIP library, so let's make that better, and then we can create an open source SIP capable VoIP-phone that could interoperate with this system as well as others.
Other uses for SIP that could/should happen IMO are (starting a session of) multi-player games and messaging, conferencing software for sharing pictures, etc.
Since SIP is basically just a handshake protocol, doing all those things shouldn't be impossible. Wanna play a game of chess or go with a pal? Just initiate a SIP connection, if their end supports your game and they are available, you've got a connection. No more application specific ports to configure to get a multiuser application work.
As the name implies and the article explains, the phone uses SIP, or Session Initiation Protocol. I did some research on SIP last year and found it to be somewhat intruiging.
SIP is basically used for setting up the endpoints of a human communication channel over an IP-based network. It negotiates what kinds of communcations are supported on each end, and what protocols to use. So if a video-SIP-phone calls a regular analog phone via a SIP-PSTN proxy, the proxy would only support audio certain codecs. The calling video-SIP-phone and the proxy would negotiate to use only audio using a matching protocol and the cal would go through.
And since SIP is a protocol just like SMTP or HTTP, it is very controllable. There are dozens of SIP products popping up from SIP servers to SIP proxies... and now SIP phones. For example, you can have a SIP proxy/server be concious of where a user is logged in and re-route SIP calls to their present location. As a Java programmer, I'm looking forward to the day when I find a reason to write a SIP Servlet.
Furthermore, the latest version of Messenger in Windows XP supports SIP. I would think that this means a SIPPhone could call someone using Microsoft's Messenger on Windows XP. However, I was not able to confirm this with a breif perusal of the SIPPhone site, and they also state this only works with other SIPPhones. That may be an over-generalization to keep people from thinking it works with regular phones, or maybe they did something crazy with it.
I'm crossing my fingers that it is a generic SIP endpoint that can contact any SIP-enabled device.
Our company have several Cisco VoIP phones deployed in various departments. We even have the ability for them to interface with the PTSN through special hardware attached to a 5ESS switch.
The only thing that prevents us from doing any massive rollouts is the utter fact that price per user and the nature of data networks make the phones more subject to unusability due to network problems than a normal phone.. This is not latency issues were more worried about something like a OSPF/Firewall or something along those lines wiping out a whole department's ability to communicate.
Looks like a picture to me :-
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http://www.sipphone.com/tiki-index.php?page=Ord
I love stacking my barbecues in the shed at the end of summer - you can't beat a bit of grill on grill action.
The dialling is a bit complicated but you can set up common numbers in the router.
It also has the problem of only being able to phone other VOIP systems but for the home worker connecting to the office (that has a VOIP exchange) it would be ideal.
Zero cost phone calls to colleagues.
If you would like to give SIP a try just try out the latested X-Lite/FWD Client available
here.
Please read the FWD Quick Start Guide to you get familar with our community.
Once you have a Free World Dialup account, you can dial your friends who have a SIPphone account, by dialing **747 followed by the SIPphone number. You can also dial people on other SIP networks.
FWD now also supports the ability to place
"toll-free" calls into the US, UK and the Netherlands. more Details are availalbe: here.
At the moment there are approx. 44,000 FWD subscribers in 150+ countries.
Not quite accurate; SIP phones can call POTS lines and vice versa assuming whomever is running the SIP gateway has a land line.
My buddy's company, Vail Systems runs a completely SIP based network at this point. They are definately making calls to people using standard telephones.
Yes, but don't forget that there are still issues using SIP across NAT gateways/firewalls.
On the plus side, even though the Grandstream phones have had some flakiness issues, Grandstream has been very responsive, releasing firmware updates regularly.
They also seem to be committed to working with open source projects like Asterisk, perhaps even supporting Asterisk's IAX protocol - a replacement for SIP that DOES work (and work well) behind/across a NAT firewall.
In any case, anyone interested in this stuff really should mosy over to Asterisk's website (asterisk.org). A lot of progress is being made putting together a very powerful open-source PBX system.
Yes and no. The phone can only call other SIP devices, but there is no reason that the SIP device cannot be a gateway to the PSTN. Mr. Roberton's service includes the ability to call other sip directory networks, including Free World Dial-up.
Free World Dial-up already has the ability for USA and UK PSTN phones to call a FWD phone number (see the "3rd Party Inbound" section at http://fwd.pulver.com/index.php?section_id=78 ). In addition, the same page explains how to call USA nationwide, UK, and Neatherlands Toll Free numbers from your FWD SIP phone. Since SIPphone can call FWD, they are able to do the same.
So, maybe this is not so useless afterall... ;-)