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Questions tagged [jain-sip]

JAIN-SIP, or JSIP, is a Java specification for SIP Signaling.

Overview
The Java APIs for Integrated Networks (JAIN) is a JCP work group managing telecommunication standards. Session Initiation Protocol (SIP) is a standard communication protocol. Java and SIP together create the JAIN SIP API, a standard and powerful API for telecommunications. This idea started in 1999 with JSR 32. The reference implementation is open source, very stable, and very widely used. This API is frequently (but not necessarily) used for client-side application development. Other container-based technologies, like SIP Servlets, are simpler to use for server-side development.

Useful Links
JAIN-SIP homepage

167 questions
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Serializing/deserializing a SIPDIalog

I was trying to serialize and deserialize a gov.nist.javax.sip.stack.SIPDialog object into Cassandra. But the equals comparison on the deserialized object fails when I compare it with the original SIPDialog object I serialized. SO looks like I am…
jeera
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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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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
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PeerUnavailableException using JAIN SIP API and the NIST implementation

I'm trying to build a simple SIP application using JAIN SIP 1.2 and the NIST implementation. I'm using JavaSE1.7 with Eclipse as my IDE. The problem: I am getting javax.sip.PeerUnavailableException when trying to construct a SipStack object. My main…
Johannes Liebermann
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Why sometimes Jain Sip return "481 Call Leg/Transaction Does Not Exist" on received bye?

Using jain sip to create sip proxy app. client A -> proxy app -> client B In case A call B: A can hangup, but when B send BYE sometime Jain Sip return 481 call leg to B. When the call end in seconds, it can BYE success. But if the call end in…
chunpo
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Failure of SIP Proxy Authentication

I'm developing a SIP user agent application that connects to an Asterisk server and tries to do an outgoing call. I'm using the NIST implementation of the JAIN SIP API. When the application registers itself, a 401 (Unauthorized) response challenges…
bgh
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How to make outgoing call via JAIN-SIP to asterisk

I try to migrate my project from asterisk-ami to SIP protocol. However, I couldn't find any examples to create outgoing call via jain sip. Some documentation mentions project sip-communicator but he is outdated. About project: it's notification…
Vladislav Kysliy
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Restcomm RTP standalone usage

I'm using jain-sip as standalone application. Everything works fine for my purposes. I need to do some modification to my application - play audio file before terminating a call (something like PLAY_ANNOUNCEMENT). It would be something like this: …
Andrei Simileac
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SIP initiate call to two parties and merge

I am newbie using SIP protocol, I googled alot to find an answer for the following but unfortunately I didn't find an answer. Is it possible using SIP protocol to initiate call from a SIP server to two clients, and then merge the two calls?
antoine magdi
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Jain SIP on android

I need to implement SIP calls with JAIN-SIP api on Android. But can't find any working example. I want to download an example, set there my params (username, domain, password) and make a call.
AndrOvip
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JSIP (JAIN SIP) exchanging request URI and Route header when sending. Why? How to prevent it?

I am using JSIP (JAIN-SIP) version 1.2 (implementation build 2384). I am trying to send the following SIP message (I removed actual IP addresses in the post): INVITE sip:bob@miniims.net SIP/2.0 Via: SIP/2.0/UDP…
Balogh Ádám Software
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Android M : VoIP, JAINSIP, SipStack; inconsistent SipProvider Exception when creating a listening point with tls enabled

UPDATE 5: Gist of what is happening: The tcp listening point is created under stack initialisation. For that, we cannot use an existing object, in our case. So what i do is check if the mTcp is not null, and if it is then the listening point is…
Pararth
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SIP Callee does not get notification that call ended

I have deleted my previous question and post this updated: I have a an issue with my SIP UAC, once I received a ringing from the B2BUA on both the caller and callee, and the caller hangs up the call while the call is ringing (I send cancel request…
TacB0sS
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What would be the best implementation to detect repeating SIP message?

I've wrote a SIP UAC, and I've tried a few ways to detect and ignore repeating incoming messages from the UAS, but with every approach I tried, something went wrong, my problem is that all the messages that has to do with the same call has the same…
TacB0sS
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RTP streaming in SIP

I am new to SIP and RTP both. I have successfully managed to create a SIP call but I still dont have voice for the session. I understand that I have to create a RTP stream and send packets. But I am unable to decide where to start from. I found…
gourig
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