Paul Jacobson - Latest Comments in SIPping from the VOIP fountain

Re: SIPping from the VOIP fountain

pauljacobson — Tue, 21 Oct 2008 02:24:11 -0000

Hi there

I have installed Nimbuzz and I am testing it out from an IM
perspective. It seems pretty decent. I haven't tried voice services
using any of these services as yet so I haven't compared voice quality.

I only installed Gizmo when I posted about it so I don't know what it
was like before then. I am not very impressed with the Mac client
though (it just doesn't seem to be able to scan my contacts and import
other users).

Re: SIPping from the VOIP fountain

pauljacobson — Tue, 21 Oct 2008 02:24:11 -0000

Hi there

I have installed Nimbuzz and I am testing it out from an IM

perspective. It seems pretty decent. I haven't tried voice services

using any of these services as yet so I haven't compared voice quality.

I only installed Gizmo when I posted about it so I don't know what it

was like before then. I am not very impressed with the Mac client

though (it just doesn't seem to be able to scan my contacts and import

other users).

Re: SIPping from the VOIP fountain

symbianism — Tue, 21 Oct 2008 02:16:02 -0000

Hi Paul,

May I suggest you try another SIP/Skype/VoIP/chat app?

Nimbuzz recently released a new version, and they've beaten fring on sound quality. That wasn't too hard, because the fring server that routes your calls appears to be a bit overloaded.

fring is still good, but that's mainly because of their instant messaging features. In terms of (lack of) delays and echos and static you're better off with Nimbuzz than with fring.

I recently tried the latest versions of Nimbuzz, fring, and Talkonaut: http://symbianism.blogspot.com/search/label/VoIP

It's been a while since I tried Gizmo (early July). Did you notice any improvements in Gizmo in the past three or four months?

Cheers!

Re: SIPping from the VOIP fountain

symbianism — Tue, 21 Oct 2008 02:16:02 -0000

Re: SIPping from the VOIP fountain

pauljacobson — Mon, 20 Oct 2008 10:54:19 -0000

Thanks for that info! I love that SIP is an open standard and enables so many options.

-original message-
Subject: [pauljacobson] Re: SIPping from the VOIP fountain

Re: SIPping from the VOIP fountain

pauljacobson — Mon, 20 Oct 2008 10:54:19 -0000

Thanks for that info! I love that SIP is an open standard and enables so many options.

-original message-

Subject: [pauljacobson] Re: SIPping from the VOIP fountain

Re: SIPping from the VOIP fountain

Allen Baranov — Mon, 20 Oct 2008 09:43:04 -0000

Once a call has been connected, all VOIP systems have a similar way of pushing voice through to the other side which is pretty much -> take a piece, wrap it up in as small an envelope as possible and push it through as fast as possible.

If pieces get lost then the other side will just play silence or white noise for that split second.

Since this part of the communication is has to be very quick - there is very little security involved.

Where different systems differentiate themselves form each other is in the part where a new client is added to the network and when two clients establish a session (call).

Skype has their own propriety way of adding clients and setting up calls whereas Gizmo and the like use SIP. Most VIOP phones, such as Cisco's phones et al, use the SIP open standard.

Since SIP is an open standard there are a lot of different "tools" that use it such as VOIP Cell phones, softphones such as the XPhone, VOIP phones such as Cisco's IP phones, online VOIP services such as Gizmo, open source PABXs such as Asterisk, etc.

Putting all of these technologies together can be quite fun. (And because SIP is open and very widely supported they CAN be put together) which means you could for example:

1. Get someone to phone a number (here or in the US or another country) which is then converted to IP,
2. It gets routed to a "switch" box which knows if you are in the office or not.
3. If you are in the office then the call gets routed to the desk that you happen to have logged on to or to a softphone on your PC.
4. If you don't pick up in time then the call gets routed via wireless to your cell phone.
5. If you still don't pick up then the system guesses that you are out of the office and routes the call to your cell phone via the Internet
6. If you still don't pick up then it gets routed to your cell phone number,
7. If you still don't answer then the call gets routed back to your PABX where a person has the option to leave voice mail.
8. If you are out of the office then the call doesn't get routed to your desk phone or via wireless - it just goes straight to your cell phone via the internet.

There are about 5 different "products/services/tools" in action while this is happening and since they all talk SIP - they can all work together.

This is why SIP is so powerful and useful.

Re: SIPping from the VOIP fountain

Allen Baranov — Mon, 20 Oct 2008 09:43:04 -0000

Once a call has been connected, all VOIP systems have a similar way of pushing voice through to the other side which is pretty much -> take a piece, wrap it up in as small an envelope as possible and push it through as fast as possible.

If pieces get lost then the other side will just play silence or white noise for that split second.

Since this part of the communication is has to be very quick - there is very little security involved.

Where different systems differentiate themselves form each other is in the part where a new client is added to the network and when two clients establish a session (call).

Skype has their own propriety way of adding clients and setting up calls whereas Gizmo and the like use SIP. Most VIOP phones, such as Cisco's phones et al, use the SIP open standard.

Since SIP is an open standard there are a lot of different "tools" that use it such as VOIP Cell phones, softphones such as the XPhone, VOIP phones such as Cisco's IP phones, online VOIP services such as Gizmo, open source PABXs such as Asterisk, etc.

Putting all of these technologies together can be quite fun. (And because SIP is open and very widely supported they CAN be put together) which means you could for example:

1. Get someone to phone a number (here or in the US or another country) which is then converted to IP,

2. It gets routed to a "switch" box which knows if you are in the office or not.

3. If you are in the office then the call gets routed to the desk that you happen to have logged on to or to a softphone on your PC.

4. If you don't pick up in time then the call gets routed via wireless to your cell phone.

5. If you still don't pick up then the system guesses that you are out of the office and routes the call to your cell phone via the Internet

6. If you still don't pick up then it gets routed to your cell phone number,

7. If you still don't answer then the call gets routed back to your PABX where a person has the option to leave voice mail.

8. If you are out of the office then the call doesn't get routed to your desk phone or via wireless - it just goes straight to your cell phone via the internet.

There are about 5 different "products/services/tools" in action while this is happening and since they all talk SIP - they can all work together.

This is why SIP is so powerful and useful.