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4

Linphone should do the trick. It supports call encryption via ZRTP. NAT traversal is easy with it via specifying a STUN server (i.e. stun.ekiga.net). It's super easy to set up. However, as it is SIP based, you will need a SIP server to go through. Personally, I just use sip.linphone.org as my proxy. However, you should be able to set up SIP Witch on both ...


4

Normally, routers have a page called QoS (Quality of Service) which allows you to do exactly what you are asking for. The configuration page varies from router to router. In my Cisco router it is: . You can pick an application, or define a new one, and then assign it a varying priority. Since I do not know make and model of yours, I cannot be more specific, ...


3

I am a avid Google voice user and would recommend it. Of course this would all be based on the assumption he already uses gmail (or is willing to). You can call from either your gmail or Google voice interface, texting and VM will be handled from your Google Voice interface. In Google voice check the setting that sends calls to your Google chat and it ...


3

As per the links you provided, you must change your status to "Offline" before closing the application. Until Skype changes this behavior, it seems like your only 1-step solution.


2

Modern asterisk works same with pci card or without. Thoose recomendation valid for linux kernel <2.6.18, that is not actual in modern distro.


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

When I looked up the specs of this device they say it supports DNS SRV under RFC:2782. You can find the specs here You can find additional information about DNS SRV at RFC2782. Here is some good information using DNS SRV with a SIP implementation:


2

No. The equipment needed to run both DSL and VoIP (the modem and VoIP box) require an outside power source to function. Unlike traditional pots line VoIP and DSL cannot use the existing DC current on a circuit. Some areas offer the capability of plugging in an ordinary handset to these lines to reach emergency services in the event of a power outage. If you ...


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

NAT provides security by obscurity as a side-effect of how it works, not by design. If you are concerned about security, you should have a firewall or packet filter on your router, which is meant to provide security by design, whether or not NAT is in the mix. As far as what access this gives people external to your network, it depends on the vendor's ...


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]: ...


1

Paul's answer is ok, but better use "sip" protocol instead of "tel" otherwise QuteCom will try to call "tel:#########", because browsers pass URI into %1. Also in Windows 7 and latest version of QuteCom my path looks like this (notice slash instead of backslash): C:\Program Files (x86)\QuteCom\QuteCom.exe -c call/"%1"


1

In X-lite Softphone -> Account Settings Under "Allow this account for" Check "IM / Presence" I can't believe this took me so long to figure out but I also can't believe I couldn't find an answer to this anywhere.


1

If you can spare some money I would suggest buying a VPS somewhere where skype is not blocked (i.e. Linode VPS in London) and then if you are linux user just ssl-tunnel into your VPS and run skype through that tunnel. You can tunnel through known port (i.e. 80) so this should pass uncovered through internet provider. And nice thing about VPS - you can build ...


1

If you can send yourself an email, you can probably send yourself a text message by using an email-to-SMS gateway: http://en.wikipedia.org/wiki/List_of_SMS_gateways


1

I think we can only guess at the authors intent and target audience (and bear in mind that was written in over 5 years ago now). I'd posit that the issue is that SSH is designed to handle TCP streams rather then UDP. Although I have found reference to tunneling UDP over SSH it appears to be a more complex hack - probably not something ideal for a VOIP ...


1

I would view it the other way around: The first difference is in the handset or the "phone" itself. Is the handset designed specifically for IP telephony? If not, then you need voice-enabled routers. Otherwise, you need handsets and routers which are compatible with one another. POTS = plain old telephone system. If you take an old phone off of POTS ...


1

This is, of course, BS. Voice over IP is general term referring to everything from Skype to SIP. Indeed, IP Telephony, although probably a marketing term by Cisco, is more of a synonym.


1

Assuming a direct line to your DSLAM, this might work. Still, it depends on the network operator to supply emergency power to the DSLAM, which probably requires quite a lot of energy. Long story short: VoIP will not be usable in case of a disaster.


1

As there are many settings that could influence the quality of VoIP, I recommend comparing all settings with those listed in the article : How to configure your router for blazing speeds. You should especially disable all unnecessary WiFi modes and protocols, for example "b+g WiFi mode" if you only have "n" devices, or 5GHz if you only have 2.4 GHz devices. ...


1

Do you want to develop the App yourself? I would choose one of the many apps already available. First you need to decide what protocol you want to use. Users could just use Skype to communcate from Skype to Skype. Downside/Upside is you don't need a server of your own. If you want to setup your own server you could go for the widely used SIP-protocol. Here ...


1

There is full documentation about ports availible. It is not clear why you think it not availble. Asterisk use port depend of technology/channel type used For sip protocol asterisk use port described in sip.conf(default 5060) and for rtp data use ports described in rtp.conf(default 10000-20000). Usually sip use only udp port. TCP-sip can also use tcp port. ...


1

Have you checked that skype isn't still running on your phone? It happens quite often on my Galaxy S3 where I will exit the app but it still runs in the background without the icon in the taskbar and no notifications. Usually when this happens I just go to the app info and force close it.



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