Questions tagged [voip]

Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services — voice, fax, SMS, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream.

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. The codec used is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.

http://en.wikipedia.org/wiki/Voice_over_IP

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Cross-platform VOIP + IM client and Linux server

Right now, a group of our employees is using Skype for group chats and conference calls. They're not using telephone numbers, just Skype user accounts. I'd like to set up a server that will provide similar functionality that is only accessible…
Jim Hunziker
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Causes of RTP jitter at the server

Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was rather surprised to see Wireshark reporting a huge…
miken32
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Is it possible or advisable to virtualize a PBX system? How would one go about this?

I'm totally new to the world of VoIP and we are looking to move from our current provider to a solution we host ourselves, mainly because the current service is so unreliable. Unfortunately I know basically nothing about VoIP and what is necessary…
tacos_tacos_tacos
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VoIP PBX, custom build or appliance?

I'm looking for advices regarding VoIP products. I need to build an in-house VoIP PBX for one of my company's office, and while I'm fairly sure we will go with Asterisk (still leaving the door open to OpenSIPS), I'm not certain I want to build it…
Julien Vehent
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Simple Voip server for Ubuntu

I have a network that connects 2 households and would like to setup a simple VoIP server so that I can call between the houses. It needs to be for Ubuntu, I have used the free 3CX system and while it works brilliantly it doesn't support Linux and I…
RC1140
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Ways to monitor SIP termination on an asterisk server

I have a nagios setup which ensures that SIP is responsive on my Asterisk server, that's straight forward. My question is, what kind of possibilities are there that the Asterisk server can actually terminate properly with the termination provider?…
imaginative
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Bringing people into an Asterisk conference call

I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a…
Harley
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Unable to call through asterisk

I want to create a voip service.I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other…
sk
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How should I structure a VoIP phone system?

I'm going to be setting up VoIP for a small business that is expected to grow rapidly. I don't really understand much about VoIP other than it's a telephone system that works over a regular network (intranet/internet) rather than over the POTS. My…
Max Schmeling
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Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the current call centre. As such, I'd need to be able to…
jonathanserafini
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Is it possible for Wireshark to drop packets purposely?

I would like to test something like VoIP. I would like to test with some "artificial packet loss". Is Wireshark able to do this? Or is there any good solutions?
Harold Chan
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Using an extension to block a caller

I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I want to dial some number…
Trewq
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VoIP phone recommendation with good speakerphone under $300

I'm looking for about a dozen VoIP phones for a small office. Criteria: Not Polycom (terrible support) Under $300 USD Good speakerphone quality Any brands/models you would recommend based on your experience?
Justin
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What bandwidth would be required for 55 VOIP Lines And what type of Internet connection

In a call centre environment, what bandwidth would be required for 55 lines? This would be on SIP protocol, would we be best to use G729 codec as i know that our sip provider supports this? What type of internet connection would be best for such a…
Gary B2312321321
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Register asterisk to sip trunk

I want to register my asterisk server to a SIP trunk. I have added following piece of code in my sip.conf and extensions.conf sip.conf [general] register => myusername:mypassword@sip.flowroute.com allow=ulaw [flowroute] ; keep…
bluewhale
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