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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

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 ...


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

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 ...


4

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.


4

Ekiga ? formaly known ans Gnome meeting.


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

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, ...


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

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 ...


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

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

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.


3

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. ...


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

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 ...


3

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

There is not an unique way to do it. This is one way: I defined my extensions starting with anything but 9, and use 9 to signal that it's a PSTN call. In extensions.conf add something like this: exten => _9X.,1,NoOp(Call to PSTN) exten => _9X.,n,Dial(DAHDI/g0/${EXTEN:1}) Put it on the same context where you put the dialplan for the extensions. ...


2

I run wireshark and I see that I'm sending it correctly but my router changes the source IP that I spoofed to the public IP address and I can't find anyway to bypass this. Your NAT-enabled router is doing its job here. If your router has a non-stock firmware like DD-WRT or OpenWRT it's probably possible with some iptables commands to tell your router ...


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

Using Fedora 11 I was just able to get Pidgin to do the chat-thing. At rpm.pbone.net I was able to find the plugins I needed for Fedora 11. I first downloaded pidgin-snipe. And when I attempted to install it I was missing the dependency purple-sipe: ...


2

Remember that just using TOR for SIP (and not for actual data) isn't good for anonymizing your connection. Using voip over TOR is definitely foolish, because TOR is slow by it's nature. Your traffic will go through multiple nodes, some being normal home broadband connections. Each connection adds latency, in my experience with rather much (latencies over ...


2

I made my thesis this year, whose subject was SIP over Tor. Soon you will be able to read a paper we prepare about this. To the point, I made measurements of delay when using SIP over Tor and I could say the results where quite interesting. Both caller's and callee's servers should use Tor, so if they have a really fast internet connection, then delay is ...


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

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

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.



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