p2p-sip | Open source peer-to-peer Internet telephony | TCP library

 by   theintencity Python Version: Current License: Non-SPDX

kandi X-RAY | p2p-sip Summary

kandi X-RAY | p2p-sip Summary

p2p-sip is a Python library typically used in Networking, TCP applications. p2p-sip has no bugs, it has no vulnerabilities and it has low support. However p2p-sip build file is not available and it has a Non-SPDX License. You can download it from GitHub.

Open source peer-to-peer Internet telephony (P2P-SIP) software in Python
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            kandi-support Support

              p2p-sip has a low active ecosystem.
              It has 84 star(s) with 55 fork(s). There are 16 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              There are 14 open issues and 4 have been closed. On average issues are closed in 1 days. There are 1 open pull requests and 0 closed requests.
              It has a neutral sentiment in the developer community.
              The latest version of p2p-sip is current.

            kandi-Quality Quality

              p2p-sip has 0 bugs and 0 code smells.

            kandi-Security Security

              p2p-sip has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.
              p2p-sip code analysis shows 0 unresolved vulnerabilities.
              There are 0 security hotspots that need review.

            kandi-License License

              p2p-sip has a Non-SPDX License.
              Non-SPDX licenses can be open source with a non SPDX compliant license, or non open source licenses, and you need to review them closely before use.

            kandi-Reuse Reuse

              p2p-sip releases are not available. You will need to build from source code and install.
              p2p-sip has no build file. You will be need to create the build yourself to build the component from source.
              Installation instructions, examples and code snippets are available.
              p2p-sip saves you 11392 person hours of effort in developing the same functionality from scratch.
              It has 23055 lines of code, 1237 functions and 61 files.
              It has high code complexity. Code complexity directly impacts maintainability of the code.

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            Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of p2p-sip
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            p2p-sip Key Features

            No Key Features are available at this moment for p2p-sip.

            p2p-sip Examples and Code Snippets

            No Code Snippets are available at this moment for p2p-sip.

            Community Discussions

            QUESTION

            How to set a write deadline for GoLang bufio.Writer?
            Asked 2022-Mar-22 at 14:14

            I am using buffio.Writer in GoLang as follows.

            ...

            ANSWER

            Answered 2022-Mar-22 at 14:14

            There are two cases to note here.

            If you want to have per write() deadline, then its not possible to use buffering. When you use buffering, then the actual write() is triggered when the buffer is full. So technically its not possible to know when your write() is completed. In this case, you are essentially using conn.write() and you can use conn.SetWriteDeadline(time.Now().Add(n * time.Second)).

            In the second case, as @icza has mentioned in the comment, you can set the deadline in the underlying conn object, and the buffio.writer() wrapper will adhere to this rule. While this is semantically correct, it doesn't provide the networking abstraction you want.

            Source https://stackoverflow.com/questions/71572909

            QUESTION

            What is the proper way for me to handle network access for a device
            Asked 2022-Mar-12 at 18:38

            I am currently working on a blazor server project which will display information from modbus tcp/ip devices. I have a class called "DeviceModel" which models a Modbus device. A simplified example is shown below.

            ...

            ANSWER

            Answered 2022-Mar-12 at 18:38

            Or does it make more sense to keep NetworkAccess and DeviceModel separate?

            As single responsibility principle of SOLID says:

            The single-responsibility principle (SRP) is a computer-programming principle that states that every module, class or function in a computer program should have responsibility over a single part of that program's functionality, and it should encapsulate that part. All of that module, class or function's services should be narrowly aligned with that responsibility.

            Read more about single responsibility principle of SOLID here.

            So making separate method dev.ResetAlarm1() in Device class is more preferable for me.

            It is hard to say whether my refactoring code is appropriate to you, but I tried to do my best:

            Source https://stackoverflow.com/questions/71441357

            QUESTION

            Any way to know how many bytes will be sent on TCP before sending?
            Asked 2022-Feb-21 at 00:28

            I'm aware that the ::send within a Linux TCP server can limit the sending of the payload such that ::send needs to be called multiple times until the entire payload is sent.

            i.e. Payload is 1024 bytes

            sent_bytes = ::send(fd, ...) where sent_bytes is only 256 bytes so this needs to be called again.

            Is there any way to know exactly how many bytes can be sent before sending? If the socket will allow for the entire message, or that the message will be fragmented and by how much?

            Example Case

            2 messages are sent to the same socket by different threads at the same time on the same tcp client via ::send(). In some cases where messages are large multiple calls to ::send() are required as not all the bytes are sent at the initial call. Thus, go with the loop solution until all the bytes are sent. The loop is mutexed so can be seen as thread safe, so each thread has to perform the sending after the other. But, my worry is that beacuse Tcp is a stream the client will receive fragments of each message and I was thinking that adding framing to each message I could rebuild the message on the client side, if I knew how many bytes are sent at a time.

            Although the call to ::send() is done sequentially, is the any chance that the byte stream is still mixed?

            Effectively, could this happen:

            • Server Side
              • Message 1: "CiaoCiao"
              • Message 2: "HelloThere"
            • Client Side
              • Received Message: "CiaoHelloCiaoThere"
            ...

            ANSWER

            Answered 2022-Feb-21 at 00:28

            Although the call to ::send() is done sequentially, is the any chance that the byte stream is still mixed?

            Of course. Not only there's a chance of that, it is pretty much going to be a certainty, at one point or another. It's going to happen at one point. Guaranteed.

            sent to the same socket by different threads

            It will be necessary to handle the synchronization at this level, by employing a mutex that each thread locks before sending its message and unlocking it only after the entire message is sent.

            It goes without sending that this leaves open a possibility that a blocked/hung socket will result in a single thread locking this mutex for an excessive amount of time, until the socket times out and your execution thread ends up dealing with a failed send() or write(), in whatever fashion it is already doing now (you are, of course, checking the return value from send/write, and handling the exception conditions appropriately).

            There is no single, cookie-cutter, paint-by-numbers, solution to this that works in every situation, in every program, that needs to do something like this. Each eventual solution needs to be tailored based on each program's unique requirements and purpose. Just one possibility would be a dedicated execution thread that handles all socket input/output, and all your other execution threads sending their messages to the socket thread, instead of writing to the socket directly. This would avoid having all execution thread wedged by a hung socket, at expense of grown memory, that's holding all unsent data.

            But that's just one possible approach. The number of possible, alternative solutions has no limit. You will need to figure out which logic/algorithm based solution will work best for your specific program. There is no operating system/kernel level indication that will give you any kind of a guarantee as to the amount of a send() or write() call on a socket will accept.

            Source https://stackoverflow.com/questions/71199777

            QUESTION

            Mimic ZeroMQ SUB-Socket (in a PUB/SUB system) with e.g. Socket/WebSocket in Flutter
            Asked 2022-Feb-17 at 19:36

            Community,

            I want to use/subscribe a to a pub-socket on a server that implements ZeroMQ (https://zeromq.org/)

            My final product will be a flutter app. It must be running on Android/iOS/Windows/MacOS/Linux/Web. So I'm really careful with the plugin-choice. I do not want to burden myself with an intense amount of platform-specific code, neither do I want to be dependent on plugins that might break under certain conditions on each platform.

            I know that there is a ZeroMQ-Plugin, but it has some Unresolved Issues in terms of operability on different platforms. Also I tried to run it on different Windows-machines and it only worked in about 25% of the cases.

            Here's the fundamental network-communication between App and Server (see image below).

            Is it possible to connect to a ZeroMQ-Publisher-Socket WITHOUT implementing or depending on the C++ compiled file of ZeroMQ? I'm thinking of a Socket or WebSocket, but I'm not even sure if it's technically possible (protocol etc), as I think that ZeroMQ uses it's own protocoll (please verify).

            Can I subscribe to a ZeroMQ-Publisher-Socket with Sockets or WebSockets in Flutter? If yes, how? Are there alternatives?

            dartzmq/install

            Best regards

            ...

            ANSWER

            Answered 2022-Feb-17 at 19:36

            Q1 :
            "Is it possible to connect to a ZeroMQ-Publisher-Socket WITHOUT implementing or depending on the C++ compiled file of ZeroMQ?"

            A1 :
            Yes, it is. It is quite enough to re-implement the published ZeroMQ ZMTP RFC-s relevant for the use-case & your code is granted to become interoperable, irrespective of the implementation language / deployment ecosystem, if it meets all the ZMTP RFC-s' mandatory requirements. So it is doable.

            Q2 :
            "... ZeroMQ uses it's own protocoll (please verify)."

            A2 :
            No, in the sense of OSI-ISO-L2/L3 stack.
            Yes, in the sense of higher layer application-driven protocols, where the ZMTP RFC-s apply for the most of the ZeroMQ Scalable Formal Communication Patterns' Archetypes ( may read more on ZeroMQ sockets are not sockets as you know them ), yet there are also tools to interface with O/S plain-sockets' fd-s, where needed. Still A1 applies here.

            Q3 :
            "Can I subscribe to a ZeroMQ-Publisher-Socket with ...? If yes, how?"

            A3 :
            Yes, it possible when your code follows the published ZMTP RFC-s. Implement all ZMTP RFC-s' mandatory properties & you are granted an interoperability with any other, ZeroMQ-ZMTP-RFC-s' compliant, node.

            Q4 :
            "Are there alternatives?"

            A4 :
            Yes, if your design can extend the Server-side, adding another AccessPoint-s there, using ZMQ_STREAM Scalable Formal Communication Archetype there, may reduce your Flutter-side scope of ZMTP RFC-s needed, as interfacing to native plain-socket will be the only one to handle and the "functionality gap" thereof can be handled on the Server-side of the link ( easily handling all the subscription management & message filtering, that must meet the ZeroMQ ZMTP RFC-s, so why not tandem it inside the Server-side before connecting the down-stream to Flutter App - smart, isn't it? )

            Source https://stackoverflow.com/questions/71127181

            QUESTION

            Get cwnd of my TCP connection from a program
            Asked 2022-Jan-28 at 17:44

            I am creating a TCP connection from my linux program with boost.asio. I wonder how do I get the value of its congestion window (cwnd) from the program? The only way I know of is to parse /proc/net/tcp, but this does not feel right. I'd rather use a dedicated syscall to get this info.

            A solution to a similar question (How to monitor cwnd and ssthresh values for a TCP connection?) suggests using TCP Probe, but it feels even less appealing.

            So what is the best way to get the value of cwnd?

            ...

            ANSWER

            Answered 2022-Jan-21 at 17:00

            I did this with netlink and INET_DIAG-sockets based on this helpful example: https://github.com/kristrev/inet-diag-example

            Source https://stackoverflow.com/questions/70743339

            QUESTION

            Make reverse TCP connection accept any amount of connections (like a normal TCP server)
            Asked 2022-Jan-11 at 18:24

            I'm trying to create a reverse proxy to a CONNECT-based HTTP proxy. The user who wants to use the proxy just treats machine A as an HTTP proxy. It works the following way:

            1. machine B opens a TCP socket to machine A.
            2. On machine A, a TCP socket is exposed on a port and all the incoming data is tunneled to machine B (io.Copy).
            3. On machine B, all the data is tunneled to the local HTTP server and the socket to machine A.

            Essentially this is a reverse-proxy behind an HTTP proxy. The reason it's this complex is because the HTTP proxy is behind NAT (on machine B) and therefore not accessible directly. The use case is being able to host an HTTP proxy behind a NAT.

            Machine A tunnel (Go):

            ...

            ANSWER

            Answered 2022-Jan-10 at 19:54

            QUESTION

            Boost TCP client to connect to multiple servers
            Asked 2021-Dec-14 at 14:47

            I want my TCP client to connect to multiple servers(each server has a separate IP and port).
            I am using async_connect. I can successfully connect to different servers but the read/write fails since the server's corresponding tcp::socket object is not available.
            Can you please suggest how I could store each server's socket in some data structure? I tried saving the IP, socket to a std::map, but the first server's socket object is not available in memory and the app crashes.
            I tried making the socket static, but it does not help either.

            Please help me!!

            Also, I hope I am logically correct in making a single TCP client connect to 2 different servers. I am sharing below the simplified header & cpp file.

            ...

            ANSWER

            Answered 2021-Dec-14 at 12:00

            You seem to know your problem: the socket object is unavailable. That's 100% by choice. You chose to make it static, of course there will be only one instance.

            Also, I hope I am logically correct in making a single TCP client connect to 2 different servers.

            It sounds wrong to me. You can redefine "client" to mean something having multiple TCP connections. In that case at the very minimum you expect a container of tcp::socket objects to hold those (or, you know, a Connection object that contains the tcp::socket.

            BONUS: Demo

            For fun and glory, here's what I think you should be looking for.

            Notes:

            • no more new, delete
            • no more void*, reinterpret casts (!!!)
            • less manual buffer sizing/handling
            • no more bind
            • buffer lifetimes are guaranteed for the corresponding async operations
            • message queues per connection
            • connections are on a strand for proper synchronized access to shared state in multi-threading environments
            • I added in a connection max idle time timeout; it also limits the time taken for any async operation (connect/write). I assumed you wanted something like this because (a) it's common (b) there was an unused deadline_timer in your question code

            Note the technique of using shared pointers to have Comm manage its own lifetime. Note also that _socket and _outbox are owned by the individual Comm instance.

            Live On Coliru

            Source https://stackoverflow.com/questions/70345279

            QUESTION

            Mininet script sending traffic from virtual machine's IP instead of host machines'
            Asked 2021-Dec-05 at 11:04

            In a python3/mininet script I have a tested, valid dictionary of host machines and their IP addresses. For each of the keys - by iterating dictOfAllHostsAndIPs.keys() - I execute a script on each emulated host's terminal

            ...

            ANSWER

            Answered 2021-Dec-05 at 11:04

            I think I see what is going on in the source, but i have not run the framework to confirm it.

            It looks like mininet inatalls a NAT rule for every node:

            Source https://stackoverflow.com/questions/69469784

            QUESTION

            Floating IPs usage on Digital Ocean
            Asked 2021-Nov-27 at 00:12

            I am looking for a basic thing yet I have not found not even a single good documentation on getting it done.

            I want to allocate a floating IP, then associate it to a network interface of a droplet other than eth0. The reason is I want to have the ability to very easily switch from one IP to the other with a programming language.

            In a few words, I want to be able to do these two commands and both should provide a different response.

            ...

            ANSWER

            Answered 2021-Nov-27 at 00:12

            In the cloud (AWS. GCP etc.) ARP is emulated by the virtual network layer, meaning that only IPs assigned to VMs by the cloud platform can be resolved. Most of the L2 failover protocols do break for that reason. Even if ARP worked,the IP allocation process for these IPs (often called “floating IPs”) would not integrate with the virtual network in a standard way, so your OS can't just "grab" the IP using ARP and route the packets to itself.

            I have not personally done this on Digital Ocean, but I assume that you can call the cloud's proprietary API to do this functionality if you would like to go this route.

            See this link on GCP about floating IPs and their implementation. Hope this is helpful.

            Here's an idea that needs to be tested:

            • Let's say you have Node1(10.1.1.1/24) and Node2(10.1.1.2/24)
            • Create a loopback interface on both VMs and set the same IP address for both like (10.2.1.1/32)
            • Start a heartbeat send/receive between them
            • When NodeA starts it automatically makes an API call to create a route for 10.2.1.1/32 and points to itself with preference 2
            • When NodeB starts it automatically makes an API call to create a route for 10.2.1.1/32 and points to itself with preference 1
            • The nodes could monitor each other to withdraw the static routes if the other fails. Ideally you would need a 3rd node to reach quorum and prevent split brain scenarios, but you get the idea right?

            Source https://stackoverflow.com/questions/69919561

            QUESTION

            Julia ZMQ - connecting to other WebSockets produces StateError
            Asked 2021-Sep-27 at 07:59

            I am trying to use ZMQ to connect many publishers to one subscriber (python). This is one such publisher (I use connect instead of bind because the subscriber binds). The code works fine until I unblock the commented code below.

            I then receive this error on Windows:

            ...

            ANSWER

            Answered 2021-Sep-27 at 07:59

            This seems to be at least in part a bug (or difficult to understand behavior) so I suggest you create an issue on the repo. Perhaps it's related to: Test Error: Assertion failed: Socket operation on non-socket #147.

            However, we can do our best to try to understand what's gone wrong and perhaps find a workaround. Since ZMQ.jl uses libzmq to handle sockets on a low level it might interfere with Julia's handling of file descriptors, we may have a race condition. Let's test that hypothesis by modifying your code a bit:

            Source https://stackoverflow.com/questions/69336268

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            No vulnerabilities reported

            Install p2p-sip

            Please use the git clone to get the sources. Alternatively, you can download the latest tar/gzip archive from the download page. Or you can browse the source code with documentation extracted from the corresponding specifications online here. The annotated source code for web view is generated using the included tool htmlify.py. The downloaded archive will create the p2p-sip/ directory containing the source code in Python. The sub-directory src/ contains the source code packages app, std, tools and external. The std package contains implementation of various IETF RFCs and Internet-drafts such as RFC 3261, RFC 3550, RFC 2617, etc. The app package contains the applications such as SIP client (voip.py), DHT using Bamboo (dht.py), etc. Some of the other modules are not yet completed. You will need Python 2.5 or higher (but not Python 3.x) to run the software. Set your python path and test a couple of modules such as voip and dhtgui before modifying the source code for your needs. You need to set the PYTHONPATH environment variable before testing these modules. For example, you can do the following to test the voip module. It performs certain SIP registration, call and instant message test with the iptel.org server. Each module comes with a simple test case to test that module. I will upload more example applications built using these basic modules as demanded. The following command launches a test user interface for the P2P module. It depends on wxPython for user interface functions. This launches a user interface with a DHT circle. You can click near the center of the circle to add a new node. Alternatively, to launch the P2P-SIP node, use the p2psip.py module as follows. The first node should be launched with -s option to become a super-node, and all subsequent ones should be launched without -s to join this P2P network. The first node listens on SIP port 5062. If you use -d option then you can also see the P2P messages exchanged between these nodes. If you wish to run the P2P network across multiple IP networks, you will need to re-configure the boot-strap node because the multicast discovery typically works only within the same IP network. If you wish to test P2P-SIP using X-lite please use the following X-lite v3 configuration. In preferences/options under "Account" tab, select "Domain proxy" and set the proxy address to be the boot strap server on port 5062, or one of the other server with correct port, e.g., 127.0.0.1:5062. In "Voicemail" tab, uncheck everything to avoid sending unnecessary voicemail related messages to P2P-SIP nodes. In "Topology" tab, under "Firewall traversal", the "IP address" is set to "Use local IP address". "STUN server" is set to "Use specified server" and the address is left blank. Uncheck all other boxes and set "Use Xtunnels" to never. In "Presence" tab, the "Mode" is peer-to-peer. All other values are left as default. In "Advanced" tab, make sure to uncheck the "send SIP keep alive messages". In fact, the only checked box is the "use rport". All other values are left as default. Beyond these examples, feel free to explore the source code and learn more about various modules.

            Support

            If you have patch for a bug-fix or a feature, feel free to send me the patch to the support group. 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. Please join the support group if you want to contribute or hear about the project announcements. If you are a project student and would like to do a project in SIP, P2P or P2P-SIP, you are welcome to use this software. I will be happy to assist you in mentoring your project in my free time. One of the main objective of the project is to help the student developers understand the existing protocols for peer-to-peer and real-time communication. Thus we encourage developers to take a look at the source code and how it reflects the specifications of various protocols. The Python programming language allows us to write compact and concise software in a way that provides pseudo-code to validate the specifications. You can browse the source code with embedded comments from corresponding specifications later on this page. All individual contributors who would like to add a feature, fix a bug or modify the source code in any way, are welcome! Please follow the programming style and send your modified source code to us, and we will add it after review. All submitted code must be released under GNU/GPL, but the original author retains the Copyright. Notice: The past and current owner of this project is Kundan Singh. The owners of the project reserve all the rights to source code. All the contributors and committers automatically and implicitly assign all the rights to the owners by contributing to this project. The rights assignment implies that the owners reserve all the rights to publish or distribute the sources or binaries elsewhere under another license for commercial or non-commercial use. Irrespective of the rights, this project will continue to remain open source.
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