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Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

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Voip using sip from android app to web app

I want to integrate voice and video call feature in my android as well as web app using sip.The scenario is like the end user can call from android app or web app to my support team which would be on web app.The scenario is similar to a call center…
Rajesh Gosemath
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IOS9 voip communication

In iOS9 apple deprecated setKeepAliveTimeout and recommend UIRemoteNotification with VOIP Type available in PushKit for VOIP calling. Ok I can imagine that I will write server application for sending these notification. 1) But what-if i need…
Jakub Průša
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Making Sip call in android

I am using ngn stack library to make sip calls. If I test on local server when I am connected to WI-FI, it works very fine with audio as well as video. But, on original server I am unable to make call, even there is no request being hit on server…
Mansuu....
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Unexpected end of call when app is started using Web Trigger

I've created an RVD app enabling Web Trigger. One of the modules of this app contains a collect step. When I start the app using Web Trigger, the call ends few seconds after say the prompt message of the collect step, while I'm still dialing some…
ghjansen
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SIP Redirect via Proxy (SIP.js)

I'm trying to create a minimal SIP Proxy that serves one purpose: redirects requests to another domain. The catch is the domain I'm redirecting to requires authorization so I assume I need to rewrite some SIP attributes since SIP authorization is…
Dan
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Asterisk as WebRTC MCU and External Authentication

I have a website with an integrated and customizable chat system ( based on XMPP ). I have tried many WebRTC framework like licode, muaz-khan's, jinja's plugin, OpenTok for WebRTC and other, but neither of them had enough reliability ( especially on…
M4rk
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Any Android Intent to make SIP Call?

Can any body suggest me, any Android Intent to make SIP Call? or even third party framework/lib/app, which has the facility to be invoked using an intent and some parameters will be fine. Kindly Note: Not regular phone call, needed intent for…
Ganesh K
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Dial -Out with Asterisk Sip - circuit is busy error

Okey İ handled my problem, Problem is provider. it is rejected my request! All problem provider that means trunk! I have a asterisk server 1.6 and a trunk. i tried to call my cell phone on trunk(provider) when i call my cell phone it gives me : --…
Yasin Caner
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difference between SIP Registration and SIP Login

We need to developed SIP Client, and have one basic doubt, we got SIP UserId , password and server detail and would like to know few things, What is difference between SIP Registration & SIP Log in , I believe, there is no something like SIP…
Amitg2k12
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ERROR:mi_fifo:mi_fifo_check: security: fifo_check: inode/dev number differ: (/tmp/opensips_fifo)

I am new to opensips and have installed it a few days ago. I have got it to make calls. But i am facing a problem with mi_fifo module. It is giving the following error ERROR:mi_fifo:mi_create_fifo: fifo_write did not open: Bad file …
Shr'Ma Dexterity Ratnesh
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Behavior of sip INVITE

Good day All, I understand that when I send a SIP INVITE and do not receive a 200 OK, it will keep sending INVITES at regular duration (until sip timeout). However, If I have received a 100 Trying for the 1st INVITE and no 200 OK (still waiting for…
Bashab
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SIP tool to generate test traffic and measure latency

I'm looking for a tool (preferably open source) which could generate test traffic towards a SIP server, test traffic could be SIP INVITE/OPTIONS ping and verify the response from SIP server. I also need the tool to provide some stats on the response…
Nitesh
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Why is dynamic real time not recommended as per asterisk?

In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like…
user3705456
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first test in Kamailio

I have just installed Kamailio SIP Server followed instructions on official site. Later I've started the server for listening SIP messages and added "test" user. So now the tutorial is ended up and i have no idea how i can test whether it works…
Volodymyr
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Kamailio '403 Not Relaying' when default port changed

Hello fellow developers... We have been testing Kamailio for a week and it is working great... But some of the our friends reported that they can't connect to our server using their mobile Internet... and it seems default SIP port is blocked by…
JavaMachine
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