Flash Media Server VS Sip Server: Difference for Voip
When using Flash Media Server and when using Sip Server ? What is the difference ? What would you recommend ? . Thanks ;) ...
When using Flash Media Server and when using Sip Server ? What is the difference ? What would you recommend ? . Thanks ;) ...
I want to create an application for ONLY Pc-to-Pc call. Do I need server like asterisk or SIP ? How can I get start ? I will use P2P architecture. ...
A requirement for the SIP PBX I created for my company was to record all calls passing through it. I solved it by forcing all SIP message to pass through the PBX and to modify the SDP body so the stream passes through it and gets recorded. It works well. I recently found out that this is not allowed. Is there any other way to implem...
I want to implement the SIP protocol in java and would want to be able to create different clients (5 or more) and make them connect to a proxy server. This is all for testing purposes so I would like to be able to see well what's happening on a rather low level. The clients should first be able to communicate trough text and later on ma...
Need to find a asynchronous DNS resolver implemented in C (except Sofia Resolver) which supports DNS queries for NAPTR, SRV and A records. It would be desired to support internal caching. Any suggestions/recommendations? Currently looking at ldns which supports NAPTR, SVC and A queries. But, If I have understood correctly, it is not asyn...
Hello, 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 the network traffic with wireshark, I see a bunch of RTP traffic from my localho...
The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 ...
Hi, I'm fighting with installation SIP for Python on Mac OS X. Finally after compilation and installation when I run console form folder of SIP (locally) I can import sipconfig, but when I`m in other folder I cant - there is no module called sipconfig. My question is - Where is folder to which I have to copy modules if I want to have th...
I want to build a sip client based on SIP Communicator - the Java VoIP and Instant Messaging client. Basically I need to plug in some how and redirect VoIP to and from my application. Where is a good place to start? If this seems a bit vague, I do apologize. ...
The main requirements are: open source solution on Linux support P2P VoIP calls support presence support multicast VoIP announcements (and maybe some way of setting up such a "conference") preferably serverless (maybe the network can get split and I'd need to keep the functionality for all clients that still see each other) I tried l...
Hi, is it possible to notify a SIP client when there is an incoming call on another phone? I know that there are the SUBSCRIBE and NOTIFY commands but I have found no event package for signaling incoming calls. Background: for a SIP-capable telephony system, I would like to provide an application that displays information about the cal...
Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is ...
After sending an invite request i receive a trying answer, and immidietly after that i receive error 407 proxy authentication required. After sending ack & another invite with the proxy header i receive session progress about 1/4 of the time!! other times it just sends 407 error again & again. Any ideas? ...
I have set some custom properties for a container in WebSphere, but I can't figure out how to read those properties back. I've tried System.getProperty(...), but it's empty. I've seen references to JMX, but wondered if there was a definitive answer. ...
How can I change Max_Forward headers value in SIP request? I'm using SER and two xlite sipphones,and I want to check MaxFwd module. ...
I'm trying to get a SIP servlet chat server working, together with the textclient found here. When I use 2 clients to send messages to eachother (peer to peer), everything goes well. But when I use one or more clients together with my server, I have to wait exactly 32 seconds before the server picks up any new messages in the doMessage(...
Hey, I would like to create a SDP media field with its attributes, and there are a few things I don't understand. I've skimmed and read the relevant RFC and I understand most of what each field means, but what I don't understand is how do I derive from the Audio/Video Format of the JMF, which parameters of the format compose the rtpmap ...
Hey guys I am looking for a SIP book similar to this one on XMPP - Professional XMPP Programming with Javascript and Jquery (http://www.amazon.com/Professional-Programming-JavaScript-jQuery-Programmer/dp/0470540710) I am new to the area and any resources would be appreciated, thanks ...
Hey, I have UAC that registers to a UAS, after registration the UAS sends me an OPTIONS request, what should I answer it? only the audio media streams? Update I: Allow me to explain myself better... if I want to invite someone to a session I USE the INVITE method and negotiate the media then, for that specific session. But once I r...
I have a Titanium module and i want to use it for voice over ip using pjsip. I have changed the project settings the following way: added to the other linker flags the libraries from pjsip added to the header search paths the headers from pjsip added to the library search paths the libraries from pjsip If i do these things for a norm...