Questions tagged [sip]

The Session Initiation Protocol, defined in RFC 3261, is an application layer signalling protocol for establishing and modifying long-running relationships between two or more peers. If including this tag, be sure to also tag your question more specifically to include the language you're programming in, as well as any specific libraries being used such as [pjsip], [python-sip], etc.

The Session Initiation Protocol, defined in RFC 3261, is an application layer signalling protocol for establishing and modifying long-running relationships between two or more peers.

More information at Wikipedia article on SIP

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SIP and RTP implementation in C++

Are there reliable open source libraries written in c++ to implement SIP and RTP protocols ? If not , is it easy to implement them using boost.asio ?
Vahid18u
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Integrate SIP.js with React Native

Is there a way to integrate SIP.js (both audio and video calling are needed) with React Native? Importing the library itself is easy enough, but the issues I'm running into are: WebRTC support: instead of using the browser's WebRTC functionality…
penguinrob
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SipRegistrationListener callbacks are not getting fired

I am trying to run the SIPDemo downloaded from Google Andoid sample applications, but not able to get it running. The issue I am facing is that after I invoke SipManager.open passing it a profile and a pending intent, it should start the…
Amit
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SIP registration using Ozeki SDK not working

i'm trying to build a simple VoIP application using c# so i found that the Ozeki SDK is the simple way to do that but when i'm trying to registration SIP account using the SIPAccount class from the Ozeki SDK and my local IP it fail always and this…
Fadi
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Linux UDP Socket sendto: Operation not Permitted

I'm trying to diagnose a problem with the OpenSIPS ( a SIP proxy ) application. When sending two different UDP packets to the same IP and port, one call fails with -1 EPERM (Operation not permitted) whilst the other is fine. Both of the calls are…
Eosis
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sipdroid - Another incoming call is not display while one is continue

I have installed SIPDROID. I made three account at sip2sip. After that I have configured application using one of that account. Application works fine. Question is: If I make multiple call to the same phone using SIP is there anyway to handle all…
Priyank
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How can I make call between direct IP to IP without SIP Server

Is there any way to make call by just dialing a local IP address? Simply an IP to IP call. How can I do this? What changes should I make in pjsip code? I don't want to register in any server or VOIP provider. The call will happen only in local WiFi,…
Raj
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Why are Tags required to identify a SIP dialog?

A combination of local and remote tags along with Call-ID is used to identify a dialog. It is said that Call-ID is a unique value for a call. Why then is Call-ID not used solely to identify the dialog?
Pravi
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VoIP/SIP Soft Phone C# WPF

I need to write VoIP/SIP Soft Phone in C# using WPF interface with Audio support only. I need to have call transfer, call conference, and recording of conversations in mp3. I've looked at VoIP SDK from ABTO LLC, but it is slow at application startup…
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Getting started with a VOIP app for iOS or Android?

I'd like to create an app for iOS that does VOIP, presumably by interacting with a website. I can start with Android too. Does anyone know of any tutorials, suggestions or libraries that would be of any use. (The app would need to be rewritten for…
Moshe
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setKeepAliveTimeout is deprecated in iOS9

The current API changes for iOS9 state that -setKeepAliveTimeout:handler: is deprecated. Up to now, this was the only way that a VoIP SIP app on iOS could maintain its registration with the SIP-server. This technique is used by various apps like…
chriscap
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How can I use Twilio as a SIP trunk for my Asterisk to make and receive calls?

I have a Twilio account which has a number (let's say 8881231234), and I have Asterisk box. I'd like to use Twilio as an Asterisk trunk to be able to make calls at their rates and receive calls from that number on my Asterisk. I haven't found any…
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Incoming calls with SIP and WebRTC

I'm looking into implementing a browser-based VOIP solution that uses SIP and WebRTC and that connects to the PTSN. Basically, users give me their SIP credentials and I use WebRTC to acccess their microphone and speakers. On the page, I plan to run…
frenchie
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How to start RTP stream inside a SIP/SDP call

I've managed to set up a SIP call using the JAIN-SIP API for Java. Now I would like to stream some video once a dialog has been established. I've read that this is possible with SDP and RTP, and I've found multiple examples on how to define a…
user50685
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How to send and receive Voice Stream using RTP

I am new in SIP call using RTP, now I am trying to send and receive voice streams using RTP for sip call. I am done with connecting two emulators and able to send INVITE and INVITE-ACK using jain sip. After I got an Ack I want to start RTP for…
Satyam
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