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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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How SIP Request Call ID without IP affects server response?

I am facing a situation when my sip client sends a deregister with call ID without IP address. Call-ID: ZbTZ3VwsZoknVtlvROGGsOO8pt0hpFi. This message is causing looping of the 200 OK responses as the via header of the next register message from…
achoora
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How to design a carrier grade SIP Server?

There are some SIP Servers which handle few thousand subscribers and some other which can handle millions of subscribers with similar underlying hardware. What are the design and development factors to be considered for implementing SIP Servers…
Jay
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Not work DTMF in FreeSwitch

When an incoming call through the trunk, do not work DTMF, even in the console is no printed. In Asterisk trunk worked. DTMF type put rf2833 as Asterisk stood. On the inner rooms all works. Also if you run the applications start_dtmf, bridge, and…
Vladimir
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Kamailio concurrent calls after forked invite

I'm using a Kamailio proxy version 4.0.4. I have two UAC (Bob1, Bob2) registered with the same URI. Then a third UAC (Alice) sends a INVITE to the proxy. Both UACs receive this Invite and both accept the call. Kamailio now cancels the second (Bob2)…
Lukas Bernhard
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Removed the Ethernet cable in active SIP and RTP session

I have below SIP call scenario: A party --> Proxy --> B party Session is established and RTP is flowing, what will happen if I remove the Ethernet cable of A party ?? How the session will be terminated and what about RTP session ?
Bhushan Patil
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Runtime error with SIP-Viewer

I am trying to analyses Genesys SIP server logs with sip-viewer(Ref link) , Unfortunately i am getting the below error while running. With command prompt, Exception in thread "main" java.lang.RuntimeException: SIP Server, Version: 8.1. 101.64…
sasikals26
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Creating an Asterisk "application" to send GET requests from an endpoint via Phone Prompt

To start off, I'd like to state that this is my first dive into Asterisk related applications, and that I'm mostly a web developer. My workplace uses an MSP that installed Asterisk/FreePBX to manage our phone systems. The GUI is pretty intuitive and…
Thirk
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SIP server for sip dialer

I am a mobile app developer. I want a SIP dialer with my own server , i have a VOIPROUTE from (VOIPROUTES.COM). I want to use the sip for calling to pakistan, India and Bangladesh . i want a dialer like Platinum Dialer( you can find it on…
pasha pash
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How to implement integration using REGISTER SIP?

I have a question about integrating with a phone company (the Provider) using SIP. I have a situation: 1. A call is made to a PSTN number 2. The Provider forwards the call to a SIP Gateway 3. Twilio is the SIP Gateway, so I receive an HTTP request…
Alvis
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Android Emulator Not Sending Sip Register Request To Server

I have build android Sip Stack on using IP address 10.0.2.2 and Port number 5060 and my mobicent-jbossAS7.2 sip-server is running on port 5080 (UDP and TCP both). I have set the emulator 5554 redirect by using following commands. telnet localhost…
mubeen
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Jain Sip Stack using node.js Signalling Server

I have webrtc application and its signalling server is on node.js using socket.io. Now I am going to create android jain sip stack. I want to connect my webrtc application with jain sip application. My question is can I connect jain sip stack,…
mubeen
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How does Asterisk update/change IP address of peer (recipient) SIP account?

I have been reading about SIP and SIP Servers, especially Asterisk. Something is still confusing me: What happens if the IP address of the peer (recipient) changes? How does Asterisk know that the IP has changed? Or does the peer have to send a…
Pacemaker
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calls are made but no voice transferred to either sip client using asterisk and csipsimple

I am using csipsimple as sip client and asterisk server to set up call. Calls are made between 2 sip clients but voice is not getting transferred. Calls are made between 2 sip clients using AMI. I can give my asterisk cli log. Can anybody please…
Hitendra
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Connect sip softphone to local server using the ip address instead of 127.0.0.1

I am doing some POC on presence server using SIP servlets. I have connected softphone to the presence server using "127.0.0.1". Now, I am trying to use the ip(given by connected wifi router) instead of 127.0.0.1 to connect to the server but the…
user3275095
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Kamailio SIP Server Connection to Radius Server?

Hi i'm on a project that which we need to connect our kamailio SIP server 4.1 (installed on ubuntu and i can give SIP service from it i tried with jitsi also install radius tool for Kamailio) as a client(for AAA i guess) to Radius Server(Windows…
Alican Beydemir
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