Stack Overflow
Stack Overflow

Questions tagged [mjsip]

MjSip is a complete java-based implementation of a SIP stack. It provides in the same time the API and implementation bound together into the MjSip packages. MjSip is available open source under the terms of the GNU GPL license (General Public Licence) as published by the Free Software Foundation.

MjSip Features

MjSip includes all classes and methods for creating SIP-based applications. It implements the complete layered stack architecture as defined in RFC 3261 (Transport, Transaction, and Dialog sublayers), and is fully compliant with the standard. Moreover it includes higher level interfaces for Call Control and User Agent implementations. MjSip comes with a core package implementation that includes:

  • all standard SIP layers and components
  • various SIP extensions (already defined within IETF)
  • some useful Call Control APIs (e.g. Call-Control, UserAgent, etc.)
  • a reference implementation of some SIP systems (proxy servers and UAs)

Download

Mjsip Download

Documentation

MjSip Documentation

Development and support

The MjSip stack has been used in research activities by Dpt. of Information Engineering at University of Parma and by DIE - University of Roma “Tor Vergata” and is currently commercially exploited by CreaLab.

Popular open source projects in which mjsip has been used

19 questions
24
votes
5 answers

Android SIP stack - what to use?

I need to create use an SIP stack on Android, which will work with asterix and will give users the possibility to change codecs (i need to implement G729 and some other codecs). I'm new in this field (voice and codec), and every bit of information…
Jovan
  • 1,741
  • 4
  • 19
  • 38
20
votes
2 answers

RTP Packets are not being sent or received using mjsip

I am working on a softphone project using mjsip sip stack. Mjsip only supports g711 or PCMA/PCMU codec. I have added G729 to my project. When I build the project it shows no error. But when the phones get connected it the call gets established there…
S. M. Shahinul Islam
  • 2,780
  • 4
  • 36
  • 68
5
votes
4 answers

"SIP/2.0 488 Not acceptable here" error

I am new to MjSip and I use MjUa for creating a client. I want to connect to a asterisk server. it support G.711 but I can not config my app. I use this config: media=audio 4000 rtp/avp {audio 0 PCMU 8000 160, audio 8 PCMA 8000 160} but i still…
kamran ghiasvand
  • 836
  • 2
  • 11
  • 19
4
votes
0 answers

How can I change codec parsing system in MjSip?

I am working on a softphone project and using MjSip stack to create develop it. The core MjSip is only suporrted with PCMA/PCMU codecs. But I want to add some more codecs with it like G729, GSM, iLBC etc. In MjSip the class AudioSender.java is a…
S. M. Shahinul Islam
  • 2,780
  • 4
  • 36
  • 68
3
votes
0 answers

MJSIP and codec interfacing

Using Mjsip as the sip stack for my softphone project. There is class RtpSender.java has Method public RtpSender(OutputStream output_stream, boolean do_sync, SIPCodec sipCodec, DatagramSocket src_socket, String dest_addr, int dest_port ) { …
S. M. Shahinul Islam
  • 2,780
  • 4
  • 36
  • 68
3
votes
1 answer

MJSIP: Register android client with server: onUaRegistrationFailure; Wireshark 400/Bad Request

'Hello I try to develop a softphone with MJSIP for android. I have a simple test setup: 1 PC (Win7) with a sip phone (number 1000) 1 VM (Win7) with a sip phone (number 1001) and Freeswitch installed sip phone #1000 can call #1001 and…
B770
  • 1,272
  • 3
  • 17
  • 34
3
votes
1 answer

Programmatically calling my VoIP phone on my network

Here is the environment: There are both PCs and Cisco VoIP phones on our network. There is a phone switch on the network that allows the VoIP phones to call out, but I don't think the switch is relevant for this part of the project. This is my first…
Loduwijk
  • 1,950
  • 1
  • 16
  • 28
3
votes
2 answers

What is the difference between JAIN SIP and MJSIP?

I've been investigating various API options for making use of the SIP (Session Initiation Protocol) in Java. So far I've narrowed it down to JAIN SIP and MJSIP but I can't figure out the difference between the two. Can someone please explain why and…
user50685
2
votes
0 answers

How to solve 488 error with mjsip codec mismatch?

I am using Mjsip to build a softphone. I have integrated multiple codecs with it. For test purpose when I make call to another client which is a Portsip 2.0 (G729, ULAW, ALAW, GSM, ILBC) I get an 488 error (Not accepted here). I have integrated…
S. M. Shahinul Islam
  • 2,780
  • 4
  • 36
  • 68
2
votes
2 answers

local RTP port unreachable when using mjsip/jmf

I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose…
brian_d
  • 11,190
  • 5
  • 47
  • 72
2
votes
1 answer

Android: Send .wav to SIP-Phone via RTP (G.711 PCMU) very noisy, crackling sound based on SipDroid/MjSIP

I want to transmit(send-only) a .wav file from my android to a softphone (x-lite) so that the called person on x-lite can hear the sound of the .wav file. The scenario is as follows: Android and x-lite are both in the same WLAN and both connected to…
B770
  • 1,272
  • 3
  • 17
  • 34
1
vote
0 answers

Listening for incoming SIP messages using MjSip

I'm doing a university project in which i have to communicate with an existing server using SIP messages. I have done the part where i send the message, and i see with wireshark that the server responded, but i don't know how to receive that message…
lukarak
  • 41
  • 1
  • 3
1
vote
2 answers

web based softphone in Asterisk

We want to build the web-based softphone using SIP technology And we want to use the Asterisk as the Communication server and Java as primary technology in building the system so we have got following flash based softphone using Adobe…
ajduke
  • 4,991
  • 7
  • 36
  • 56
1
vote
1 answer

mjsip cant send message

I have implemented an android program that can receive simple messages using the MJSIP, the problem is that I would like to be able to send messages too, my class is: package org.sipdroid.sipua.ui; import java.util.Iterator; import…
maxsap
  • 2,971
  • 9
  • 44
  • 70
1
vote
1 answer

Asterisk :Don't allow to call if SIP peer is not registered

I am developing a SIP based application. In general if one registered user of asterisk calls to the user who is not registered, in this scenario call takes some minutes to hangup and user needs to wait sometime to make another call. So what I want…
Juned
  • 6,290
  • 7
  • 45
  • 93
1
2