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15

To get a deep analysis, let's look at the Ohloh entries for each item on the Wikipedia list: Empathy (Telepathy-Rakia) Jitsi (formerly SIP Communicator) Ekiga LinPhone Qutecom SFLPhone KPhone Blink: I added the project, but Ohloh can't process Darcs repositories yet. Twinkle: Can't find a publicly available repository. PhoneGaim: No longer available ...


5

So far I have only had success for text only support using Pidgin with the SIPE plugin. I'm looking into SIP Communicator (which is java based and runs on all platforms) it does have some VOIP and Video support, but I have not been able to get it to connect to my company's OCS SIP server. For Pidgin, get at least version 2.6.6 and install Pidgin-SIPE plugin ...


4

No numbers are inherently "SIP" numbers or any other type of numbers, they are just numbers. Numbers are handed out to VoIP providers and PSTN line providers alike from number vendors (such as 360 or Level 3, they do not generate the numbers themselves - though they can, however this doesn't mean they own the number, it will route only within their network ...


4

Actually VoIP is not a protocol but an umbrella term to cover voice communication over IP. So SIP is considered to be VoIP, and so is Skype. So it's called a SIP phone since it's using the SIP protocol but wouldn't work with any other VoIP protocol. Perhaps you were referring to the H.323 protocol? If so, the differences and similarities between H.323 and ...


3

Give Mikey's guide on 2nd LAN subnets a miss. Take out the firewall filter, disable 'for IP routing usage' (2nd subnet) and have a read of this guide on address mapping instead: http://www.draytek.com/user/SupportFAQDetail.php?ID=1353 Incidentally, I have never had an issue connecting a SIP phone via NAT, provided that you setup the right port forwarding ...


3

Linux command line: linphone includes a scriptable linphonec command-line version. Starting linphonec with the --pipe option will create a socket in /tmp that one can write to and read from. It fulfills all your requirements and I've tested (for my own uses) all of them successfully: Calling via SIP, transmitting DTMF sequences, recording a call to file, ...


3

Betraying my ignorance, I'm going to link to the Wikipedia list. Some you say you haven't tried: KPhone LinPhone PhoneGaim Qutecom SFLPhone SIP Communicator You might try looking at the crossplatform clients first (bolded above), as I have a hunch that they will be more developed.


3

@Nicolas is almost (but not quite) correct - SIP is a VOIP platform. It works over UDP, NOT TCP (this is an important feature and limitation). It is like Skype and Google-Voice, save that it is an open standard (unlike Skype), and designed primarily for calls (unlike Google Voice which uses Instant Messaging as the basis of the protocol). It is more ...


3

I doubt that you're still looking for an answer to this 6 months later, but here goes: The Twinkle SIP client supports executing a script when an incoming call is received and I'm sure many others do as well. To get something like this working in Twinkle, you'd write a script like the one below, then go into Edit->User Profile->Scripts and select ...


3

Right, SIP is an application protocol that runs on top of TCP. There are many different ways to transfer voice over TCP/IP, however SIP is probably the most used and understood in the industry with the maximum compatibility. If you just want to make calls from your computer, many come with the phone dialer application (not sure if it is included any more ...


3

While it's possible, it's not likely. There are a few simple ways you can tell for sure. First, do a traceroute from one client to the next to make surethe network is configured properly. If traceroute packets are leaving the LAN, somethig is seriously misconfigured. Next, start a VoIP call. After the call has started, physically pull the plug on your WAN ...


3

I don't know the definite answer, but I'd suspect yes, Skype will use your LAN. It's a trivial check and I see no reason why the Skype people wouldn't have put it in their app. Skype will contact the supernode over TCP to set up the connection. The supernode will see that both PCs are behind a NAT on your LAN, and there's certainly a port-restricted ...


2

As Wil said, SIP is an application layer protocol. SIP is used as signaling protocol in VoIP, ie to manage sessions(voice, video or file transfer) between participants. The actual media is transported using different protocols based on the need, voice packets are transferred as RTP once the SIP signalling is successfully done.


2

Ekiga (multi-platform) supports multiple SIP accounts as well as audio, video, etc.


2

A SIP trunk is a direct connection between your organization and an Internet telephony service provider. SIP trunk enables you to extend VoIP telephony beyond your organization’s firewall without the need for an IP-PSTN gateway. This simpler configuration is easier and less expensive to design, operate, maintain, and upgrade. Here is a suggested [blog]: ...


2

I found http://www.phoner.de/index_en.htm does this. Options -> External Application and then run a .bat file with something like: [InternetShortcut] URL=http://www.google.com/id=%1


2

Ekiga claims to support G729. I'd start there.


2

Take a look at Bria. I use it at work and it is great. It supports g729. http://www.counterpath.com/bria.html I also have tried Ekiga and it does not support g729, likely because the g729 license is not free.


2

I have used both pidgin and empathy to connect to our office OCS server. For pidgin I use the pidin-sipe package. For empathy, you need telepathy-haze which implements a libpurple-equivalent for telepathy, as well as purple-sipe. libpurple is the IM connection manager for pidgin, so telepathy-haze enables the telepathy framework to use pidgin plugins. ...


2

You can also use MacPorts to install pyqt, qt, and many other packages. I haven't used the pyat package, but in general I've been pleased with MacPorts. It also makes it easy to update various packages as bew versions come out.


2

I believe Blink http://icanblink.com/ will do exactly what you want. I used it for testing in the way you mentioned when I wanted to get user@domain.com type SIP addresses working on my Freeswitch server.


2

SSH protocol run on top of TCP part of the TCP/IP stack. SIP or other type of voip generally runs on top of UDP protocol. One of the significant differences between TCP and UDP is that TCP (somewhat) guarantees the delivery of each packet. UDP on the other hand does not. It is fire-and-forget. The consequence of that if there's some network hiccup TCP will ...


2

There is no difference since SIP is overwhelmingly the protocol of choice for IP telephony. However I guess you could say SIP phones are a (large) subset of VoIP phones. I would generally assume that any VoIP phone is SIP compatible.


2

According to a few studies I've read, G.711 seems to provide the best tradeoff between required bandwidth, compression delay and subjective audio quality. Cisco published an article in 2006, Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation, where they also evaluated the subjective Mean Opinion Score (MOS) over a couple of codecs, ...


2

First, a clarification. "VoIP" is a buzzword that refers to any form of Voice-over-IP, including that carried by SIP (Session Initiation Protocol) and its associated protocols. If your landline and mobile phone are configured to use different providers then you'll miss out on any emergency services (911, 119, 999, etc.) provided by the landline provider. If ...


1

You are on the right path: you will need a SIP provider, like Skype is, only with better software and extras. As I have been experiencing with them, most SIP providers are international. I've been using FreeCall for some time now. The prices are okay, almost always available, their software doesn't suck, though I almost exclusively use the web based client. ...


1

You want to go with a standard SIP provider and then you can use a standard phone connected to a voip gateway (Linksys SPA3102) or do like I have done and get a Snom 370 or other IP Phone. With either you simply connect it to your Voip account from someone like callcentric.com or Voip.ms ... I do with the pay as you go plans and I've been averaging $12 month ...


1

I posted about this just recently actually, so here was my response before: I assume you're switching due to today's business deal between Skype and Microsoft. You aren't alone! An easy one to acquire now is Google Talk. It's quite similar to Skype, with full video calling and chat. Living in the USA, Canada, and some other countries, you can even make ...


1

This is an extremely complicated question. First off, how are you incorporating POTS into a SIP call? are you hosting the gateway? if that is the case, look here for a conversion between sip and rtmp at the gateway. For client side translation, you could try something like Xuggle. Basically, RTP is the what carries the audio streams in SIP, so you need ...



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