Questions tagged [voip]

Voice over Internet Protocol (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. VoIP protocols can be further subdivided into Signalling and Media Protocols. Signalling protocols are used to establish VoIP sessions while the Media Protocols carry actual voice traffic.

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.

More information at Wikipedia page on VoIP

2751 questions
1
vote
2 answers

Android SIP call.startAudio() not working

I'm making a VoIP program for Android 2.3, so no RTP API, but it should send sound as well. When I call my Android client, everything is fine. However, when I initiate a call from the Android client, the SIP communication works, but no sound at…
LevyZidane
  • 11
  • 3
1
vote
2 answers

Delphi application to dial IP phone. [like Dialer.exe]

I am a little new to Delphi TAPI and Phone integration. So please forgive my greenness. I am trying to make a Delphi [XE2] application dial my IP phone without using the Dialer.exe I have successfully implemented the ITRequest::MakeCall method but…
Troy Harris
  • 435
  • 1
  • 14
  • 21
1
vote
1 answer

audio and video streaming - managing CPU and bandwidth

I'm looking into audio and video streaming (at the same time) between a provider and a consumer and I am wondering what are the best/common solutions to handle the balancing between audio and video when it comes to CPU and bandwidth. This is for a…
1
vote
1 answer

pjsua calls fail

I'm facing a problem with calls using pjsua. Registration on server is always successful, but most of the time I try to call to my cell phone, state of call stands CALLING for ever and nothing is really happening or I get error 406 (not acceptable).…
1
vote
3 answers

How do VoIP apps connect users with no real IP addresses together?

I want to know how two computers with VoIP software can be connected to each other and exchange voice data without having an external IP address.
user1306322
  • 8,561
  • 18
  • 61
  • 122
1
vote
1 answer

firefox plugin development in linux

Possible Duplicate: How to write a browser plugin? I'm doing my master thesis, my topic is to develop a firefox plugin for MiniSIP application(its a open source VOIP application) on linux platform. Basically i need to use the sip stack of the…
arun arun
  • 147
  • 1
  • 2
  • 5
1
vote
1 answer

iPhone restore AudioSession in Play&Record after playing youtube video

I have a problem about managing AudioSession (set as Play&Record category) interruptions in my VoIP app in iOS 5.x. When I have a call in progress, going background and starting youtube app, the audio session begin-interruption occurs and I can put…
1
vote
1 answer

WebRTC to make calls to PTSN

I'm looking at WebRTC and I'm wondering how to implement a solution where the client connects to the PTSN via SIP. It seems like a pretty new technology so I assume that this would not work on IE browsers; is this correct? Basically, I have a…
frenchie
  • 51,731
  • 109
  • 304
  • 510
1
vote
2 answers

how to let the kamailio support Edge Proxy?

As the RFC http://www.rfc-editor.org/rfc/rfc5626.txt describes , how can i extends a Edge Proxy by Kamailio ? should i write a module , or just write the configure file ? have any one already do it ,could give me some advice . thanks .
mike
  • 1,127
  • 4
  • 17
  • 34
1
vote
2 answers

asterisk load balancing using openser/opensips

I need to load balance incoming calls to asterisk. To do this, I have set up the Openser server in front of it and I loaded and configured the dispatcher modules to do so. What I want to do is that the Openser server will receive the calls and route…
Sara Ibn El Ahrache
1
vote
1 answer

What is going wrong in this SIP call? Multiple NOTIFY messages in a row before RTP established

the long string of NOTIFY messages happen after the called number answers. and after about 20-30 seconds the 503 happens and then the call connects fine with audio.
Adam Johns
  • 35,397
  • 25
  • 123
  • 176
1
vote
2 answers

pjsip iPhone missing libraries

I am working on pjsip for voice chat on iPhone. The steps I have completed from the instructions are: changed config_site.h as given. successfully ran $ cd /path/to/your/pjsip/dir $ ./configure-iphone $ make dep && make clean && make opened…
virata
  • 1,882
  • 15
  • 22
1
vote
1 answer

Wrong password issue in Samsung Galaxy SII

I am developing one SIP based application to make and receive a call.I have used shared preference to store the all the registration related data like user-name,password,context etc.This data are also stored in A2billing.And i have used Asterisk…
Juned
  • 6,290
  • 7
  • 45
  • 93
1
vote
1 answer

Is GSM6.10 audio format block or stream based?

I might be asking the wrong question, but my knowledge in this area is very limited. I'm using acmStreamConvert to convert PCM to GSM (6.10). Audio Format: 8khz, 16-bit, mono For the PCM buffer size I'm using 640 bytes (320 samples). For GSM buffer…
eselk
  • 6,764
  • 7
  • 60
  • 93
1
vote
1 answer

Audio call service provider

Does anyone know of an audio call service provider which can make a call to read something to any phone numbers in worldwide? Our use case is 1, User register our product on mobile. User may come from any country. 2, User input his phone number and…
Aplomb
  • 15
  • 5
1 2 3
99
100