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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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Setting Up Restcomm on Ubuntu 14.04 (EC2)

I am completely new to Restcomm and currently experimenting with the application. I want to setup a restcomm apps on my EC2 instance. I am aware that i can easily deploy Restcomm via the AWS Marketplace, however it only allows me to deploy on…
Jeremy
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SIP: How could two networks be connected transparently and form one logical network?

I've been learning SIP for a while now but I can't think of a way to accomplish a scenario which I have in my mind. Lets assume I have a number of SIP clients and a SIP server at home, in a private network (behind a NAT/firewall, inaccessible from…
Adam Romanek
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Genesys Platform : Get Call Details From Sip Server

I want to get Call Details from Genesys Platform SIP Server. And Genesys Platform has Platform SDK for .NET . Anybod has a SIMPLE sample code which shows how to get call details using Platform SDK for .NET [ C# ] from SIP Server? Extra Notes:…
Hippias Minor
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Twilio: making call w/o own server

Just begin to learn Twilio API. My intention is to make call directly from Android phone to any landline number. I have read QuickStart guide, build sample Android application and found that to make a call one need to have his own REST server. I…
Barmaley
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Single SIPp script for registration followed by SUBSCRIBE(tcp) wont work

I have call flow: UAC(tcp) SIPp(tcp) Regsiter (callid1) ----------------> 401 <--------------- Regsiter ----------------> 200 OK <--------------- Subscribe (callid2) --------------> 200…
Chirag Dhyani
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SIP Timers: What is use of TimerC in SIP?

RFC-3261 doesn't talk much about TimerC and when it fires. I understand the importance of TimerC in forking scenarios. Does TimerC have any role in regular SIP call?
TheLearner
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Why do we need SIP "100 Trying" response over TCP?

SIP over UDP: It's necessary to have SIP response "100 Trying" for SIP over UDP to shut the Timer-A off that would have been started by caller and hence stopping the re-transmission of the SIP message. Its really important because other responses…
TheLearner
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Creating SIP application

I am totally new to SIP and the related technologies. I am studying about what is SIP protocol and SIP transactions. Got the basic idea. I have a project to do where I have to create a very basic SIP service which will tell about the presence of…
user4175162
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Kamailio installation error on MAC 10.9, what is this error?

I am trying install Kamailio 4.1 on mac OS10.9 by command: make cfg; make all; make install But I see on Terminal, it through: mkdir -p /usr/local/etc/kamailio/ /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error…
TruongPS
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SIP Error 481 Call/Transcation Does Not Exist when transferring

Our phones are done through SIP and we were noticing this error appearing on our server. After running tests, we have determined that it only appears on inbound calls from external numbers, when we attempt to transfer that call to an internal…
Raiden616
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Call from web client to softphone Twinkle is received but gets disconnected at very next moment

I am using trying to call from web client SipML5 live demo page to a registered user at freeswitch. Now there are two problems. 1. Sometimes User 1002 is successfully connected and is able to make a call to user 1001 on twinkle. But call is…
Anurag Rana
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Load new module in Kamailio

I would like to ask, how can I load new module in Kamailio 4.1.2? Actually, I have an issue, when I tried to compile my kamaiio.cfg I've got error: root@kamailio:/usr/local/# kamailio -c kamailio.cfg loading modules under…
Patrik18
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Android SIP: Service not running for incoming sip calls

I am developing a SIP application for VOIP calls. I have modified SIPDemo as per my UI requirements. When an incoming call is triggered, my app is registered to server domain and able to receive calls, (even in the case when the app is not…
Nam
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SIP Client (Peers) - Call received failed

I have to integrate the text, voice and video chat via SIP server into my application. So that I have chosen the "Peers" from http://peers.sourceforge.net/. I have downloaded the code, registered a sip addressz(peers sip client) and call to another…
SKK
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SIP Client - JsSIP login issue in my server

I am creating an web application. In which I have to integrate the text, voice and video chat via SIP server. So that I have chosen the "JsSIP" from http://tryit.jssip.net/. I am trying to chat in the demo screen. It is working good. So I downloaded…
Manoj
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