Projects tagged ‘python’ and ‘sip’

This is a Certificate Authority to generate test certificates for various SIP applicants. These certificates are not secure and not meant for production use - they are just to help implementors be able to easily get useable certificates for testing.

What is SIPVicious tool suite?SIPVicious suite is a set of tools that can be used to audit SIP based VoIP systems. It currently consists of four tools: svmap - this is a sip scanner. Lists SIP ... [More] devices found on an IP range svwar - identifies active extensions on a PBX svcrack - an online password cracker for SIP PBX svreport - manages sessions and exports reports to various formats RequirementsPythonSIPVicious works on any system that supports python 2.4 or greater. Operating SystemIt was tested on the following systems: Linux Mac OS X Windows FreeBSD 6.2 If you use it on systems that are not mentioned here please let me know goes it goes. [Less]

The 39 Peers project aims at implementing an open-source peer-to-peer Internet telephony software using the Session Initiation Protocol (P2P-SIP) in the Python programming language. The software is ... [More] still incomplete -- especially the P2P part. The 39 peers project is developed for student developers and researchers to experiment with new ideas. It is written in Python scripting language. It supports open protocols such as IETF SIP and RTP. It is licensed under GNU/GPL license (an alternate commercial license is available as well). Visit the 39 peers project web site for more information. Quick StartSee the instructions on 39 peers download page on how to get started with the software. ContributingPlease visit the 39 peers project page about how to contribute to this project. Alternatively, if you have patch for a bug-fix or a feature, feel free to send me the patch to kundan10@gmail.com. If you plan to do significant contributions, please let me know and I will add you as a project member so that you can check in files using SVN. Join the mailing list if you want to contribute or hear about the project announcements. [Less]

SIP-RTMP gatewayThe goal of this project is to allow Flash to SIP calls and vice versa. In particular it allows multimedia calls from Flash Player to SIP network and SIP network to Flash Player. The ... [More] gateway implements translation of signaling as well as media between Flash Player's RTMP and standard SIP, SDP and RTP/RTCP. The client side API allows you or any third-party to build user interface of web-based audio and video phone that uses SIP in the back end. The user applications can be built using ActionScript for web browser as well as standalone AIR. The Gateway can run either as a server hosted by the provider, or as a local application on the client's host. Thus, this software caters to various customers and users. Further documentationI have written the source code in Python with extensive documentation in the source code itself. The code comment explains how the software works, design choices as well as overview of the client API. Please see the siprtmp.py source code for details. Feel free to browse through the client side of the code to know how the API is used in the ActionScript client. Download and browse sourceThis project shares the source code repository with the RTMP Server project. Since I own both the projects, I decided to keep a single source repository to simplify software development. Download the latest version, or get source control access to the software. I have also put the current versions of the download in the download section of this page, but I request you to get the latest version from the links above. You will also need to download the dependencies as mentioned in the Quick Start section below. Support and FeedbackIf you are a VoIP provider who needs to use this web-to-phone feature for your network, please get in touch with me on how I can help you. I can point you in the right direction from installation, provisioning, trouble shooting to building a client Flash application for your web site. If you are a developer who wants to add a new feature to the software or use the software in your project, feel free to get in touch with me. I can provide direction on which module to look at or modify for your work. My email address is mamtasingh05@gmail.com. I look forward to hearing from you! Quick StartThe software requires Python 2.5. It has an external dependency on the 39 peers p2p-sip project. Please download the latest version of the source code from the p2p-sip project page. Please follow the instructions on that site on how to install. I have provided the current instructions below, which may change later the project. bash$ tar -zxvf source-*.tgz bash$ export PYTHONPATH=p2p-sip/src:.Next, download the siprtmp source code from the rtmplite project page. Make sure to download version 3.0 or later which includes support for siprtmp module. bash$ tar -zxvf rtmplite-3.0.tgzNow that you have the p2p-sip and rtmplite directories, you can run the siprtmp module form the rtmplite directory as follows. Make sure to set the PYTHONPATH correctly to point to dependencies. bash$ cd rtmplite bash$ export PYTHONPATH=../p2p-sip/src:. bash$ python siprtmp -dThe siprtmp module takes the same command line as the rtmp module. The difference is that siprtmp module also enables the sip application for SIP-RTMP gateway service. At this point your SIP-RTMP gateway server is running on local host. TestingI will describe two test scenarios below which allow you to test the service locally without requiring an external SIP account. The testing employs the sample Video Phone client available in the rtmplite directory under videoPhone subdirectory. The rtmplite/videoPhone/bin-release folder contains a VideoPhone.html file which embeds the Flash application for the sample client. You will also need two additional software for doing the test: a SIP server and a SIP client. I use Free X-Lite SIP user agent because it also supports wide-band speex audio codec, which is required by this SIP-RTMP gateway software. I also use the sipd.py module available in the p2p-sip code I mentioned before, for the SIP server functions. You may be able to use some other SIP server or SIP user agent as long as they are in the same network as your SIP-RTMP gateway, i.e., no firewall or NAT among these. Run the SIP server in another terminal as follows since you have already installed p2p-sip code. The -d option allows you to trace various SIP messages handled by the server. bash$ cd p2p-sip/src bash$ export PYTHONPATH=app:external:. bash$ python app/sipd.py -dFirst test: In the first test, open the VideoPhone.html from browsers on two different computers (or if you don't have two different computers, perhaps from two different browsers or browser instances, so that cookies do not mess up your testing). When the Flash widget loads, it first tries grab your devices. You will need to "Allow" access to your devices and also check the "Remember" box in the Flash Player settings so that it stores your preference for this page. Next, specify your configuration information in the widget. The first is the gateway URL, which should be rtmp://localhost/sip if you are running the siprtmp gateway locally. Suppose your local IP address is 192.168.1.3 then you will use this in your SIP addresses. The next field is your SIP address. You can pick two random names such as "alice" and "bob" and then use say alice@192.168.1.3 on the first browser and bob@192.168.1.3 on the second browser. The next field is your authentication name which you can use alice and bob respectively on the two browsers. Next is the authentication password which you can put anything as our SIP server in this test does not do authentication. Then the display name can be set as say "Alice 1" and "Bob 2". When it prompts you to remember the configuration, you should check that so that you don't have to go through the whole configuration process next time for this web page. After you click next, it will try to connect to the gateway server, which in turn generates SIP registrations to the SIP server. At this point both your web clients are ready to make and receive calls. At this point, your local video should appear in the widget. In the first browser, type sip:bob@192.168.1.3 which is the SIP address of the second browser. Then click on the next button. The second browser should receive an incoming call indicated by the blinking button. Click on the blinking button to accept the call. You may click on the other button to reject the call, alternatively. Once the call is established you should be able to hear and see between these two web clients. There are a few user interface controls that allow you to switch between local video, remote video and picture-in-picture mode. When you want to terminate the call, click the appropriate button on one of the browser's widget, or simply close the browser. The other side should receive the call termination signal and close the call. Second test: In the second test, we will interoperate between this Video Phone of first user, alice, and a standard SIP user agent, X-Lite. This will be an audio call test because X-Lite does not understand the RTMP video format used by the Flash Player. Since X-Lite supports wideband Speex audio codec, we can interoperate between our gateway and X-Lite. The first step is to install and configure your X-Lite client on the second computer or as a replacement for second browser on your computer. Once installed, open the "Options" dialog box using the right-click menu, and go to the Advanced then Audio Codecs tab. Make sure "Speex Wideband" is among the enabled codecs listed on that page. Also, under the Advanced then Quality of Service tab, make sure that all the options are set to "None" otherwise X-Lite is known to cause problems in certain network conditions. Feel free to explore other settings as appropriate. Next, create a new SIP account in X-Lite using the "SIP Account" setting in the right-click menu. Add or modify the account to reflect the second user's credentials such as display name as Bob 2, user name as bob, domain as 192.168.1.3 and enable to register with domain and receive calls. Also set the outbound proxy mode to 192.168.1.3. Under the Topology tab select to use local IP address and do not discover STUN server, since all our testing is in the intra-net. Once you close the SIP Account dialog box, X-Lite will register with our SIP server on behalf of user bob. Now you can use the same procedure as before to place a call from widget of the first browser to the X-Lite user. When the phone rings, answer the phone and your should get connected between the first browser and the X-Lite phone. Only audio will work between these two clients. The widget will display blank video of the remote party. You may terminate the call either from the widget or X-Lite client. For other variations in this test, you can try initiating the call from the X-Lite client by dialing alice which is received by the first browser's widget. Similarly, you can also test canceling an outgoing call or rejecting an incoming call from either of the clients. Once you are comfortable testing the set up, you can explore further options in the gateway, the SIP server as well as the client. You may also want to build your own Flash application using the client API to connect to the gateway directly. For Flash Player to allow device access your Flash application must have a minimum dimension of 214x137. This is the dimension of the sample Video Phone application included in the software. DeploymentA real-deployment of this software will require far more testing and a few more features. Since there is no NAT and firewall traversal support in the gateway currently, you need to run the siprtmp gateway in the public Internet if you want to deploy this service. Secondly, this gateway should have direct access to the SIP server and assumes that if the SIP client is behind NAT and firewall then the SIP client or server somehow manage to traverse them, such that gateway sees the SIP side on the public Internet. Since the gateway does not implement RTMP tunneling, the connection from browser client to the gateway may not work under certain restricted firewall, such as those that block RTMP TCP port 1935 from client to the server/gateway. As I mentioned before, I would love to hear from you if you plan to use this software in your project or deployment! I may also be able to point you to the right direction on how to proceed with the deployment and help troubleshoot this software for your project. Final WordsThis software is provided with a hope to break away from the Flash Player's restrictions, to allow interoperability between Flash applications and SIP network, and to allow Flash and Flex developers to build interesting new Internet Multimedia applications using the SIP technology. As such this software is released under GNU GPL v3, and if you use this software in your project, you will also need to release the source code of your project. I believe a free software should be viral and GNU GPL gives a tool to do so. If the software does not work, you can contribute to fix it. There is no warranty or guarantee on this software. If we share our time and effort, all of us will benefit. [Less]

python daemon to allow the usage of a topcom butler 4012 with as a SIP phone. It'll probably be a python port of the code found at http://ventoso.org/luca/topcombutler4012/ made by Luca Olivetti ... [More] , meant to be used on platform where no FreePascal Compiler is available (for instance my MIPS router) [Less]

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fetion Don't use QQ again!

BSDRadiusBSDRadius is open source RADIUS server targeted for use in Voice over IP (VoIP) applications. While there are quite large number of Radius servers (including free) available in the world ... [More] , little number of them (if any) are developed with !VoIP specific needs in mind. Typical !VoIP RADIUS server should be able to take high amount of AAA requests in short time periods, handle large databases, and respond timely to prevent time-outs and request retransmission cases. Most commonly used VoIP protocols (SIP and H.323) use small number of Authentication methods (e.g. CHAP and Digest), and this can allow reduce processing overhead and code size of server. The server is released under the BSD license, which means that you are allowed to download, install, use and modify it at no charge. Get the codeTo use the latest and greatest code of BSDRadius - use Subversion with command: svn checkout https://bsdradius.googlecode.com/svn/trunk/ bsdradius [Less]

DescriptionAutomatically places calls to any SIP URI (or to any PSTN phone number, if using a VoIP gateway), waits for the call ringing, hangs up and starts over after a random number of seconds. ... [More] Written as an experiment on controlling a SIP softphone from python. Please use responsibly. RequirementsRequires python and a working installation of linphone console-only client (linphone-nox) and noah's pexpect (python-pexpect). [Less]

This is a python implementation of the Flash RTMP server with minimal support needed for real-time streaming and recording using AMF0. Advanced application service such as shared object or web API ... [More] are outside the scope. The goal is to use existing protocols and tools such as web servers as much as possible (API, progressive download). And only use this RTMP server when one needs to interoperate with Flash Player. Another objective is to keep the size small so that one can use it locally on a client machine instead of hosting on a remote server. The project was started and most of the work was done in 2007. More recently I wrote example test application and finished the server part to make it complete. If you are interested in contributing, feel free to send me a patch with your changes. If you plan to use this software in your project or want to contribute significantly in this project or its features, feel free to send me a note to mamtasingh05@gmail.com. I look forward to hearing from you! There are other open-source RTMP servers available such as rtmpy.org and osflash.org/red5. My implementation is different because it does not use the complex Twisted library as in rtmpy.org and it is pure Python based couple of files instead of hundreds of Java files of Red5. I did use AMF parsing from rtmpy.org though. Secondly my project is a much simpler version of a full Red5 server and useful only for dealing with real-time media and doesn't implement shared object or web server style applications. Quick StartThe software requires Python 2.5. After uncompressing the download or checking out the sources from SVN, run the server file with -h option to see all the command line options. bash$ tar -zxvf rtmplite-2.0.tgz bash$ cd rtmplite bash$ python rtmp -hTo start the server with default options and debug trace, run the following: bash$ python rtmp -dA test client is available in testClient directory, and can be compiled using Flex Builder. I have already put the compiled SWF file in the bin-debug directory. Open your browser and then open the testClient.html file in your browser. The user interface will allow you to connect to the server to test the connection. You can also test streams by clicking on publish or play buttons. See the README file for more information New Features/Other ProjectsFlash to SIP: Starting with version 3.0 onwards, the software includes a SIP-RTMP gateway module as well. The siprtmp project page describes the SIP-RTMP module in detail. The project depends on the SIP stack from the "39 peers" p2p-sip project. This module allows you to make Flash to SIP calls and vice-versa. With appropriate VoIP account you can also make Flash to phone or web to phone calls. Videocity: The Internet Videocity Project is another project that uses rtmplite as an RTMP server. The Videocity project aims at providing open source and free software tools to developers and system engineers to support enterprise and consumer video conferencing using ubiquitous web based Flash Player platform. The video communication is abstracted out as a city, where you own a home with several rooms, decorate your rooms with your favorite photos and videos, invite your friends and family to visit a room by handing out visiting card, or visit other people's rooms to video chat with them or to leave a video message if they are not in their home. User Comments(Aug 2009): "I tried your RTMP-SIP gateway this afternoon. It's pretty neat. Awesome. Great job. I like it more than the Red5 project. It's a very good idea to implement it with python and it's lightweight and better integrate with ..." (Nov 2009): "I was looking for a lightweight rtmp server and tried out your server at http://code.google.com/p/siprtmp/ and it seems to have been running quite well. Hats off to you. ... Thanks for the great lightweight server." If you have any feedback, criticism or comment on siprtmp or rtmplite, feel free to send them to me. [Less]