Ecosyste.ms: Awesome
An open API service indexing awesome lists of open source software.
awesome-voip
🤙Learning VoIP, RTP, pjsip and SIP
https://github.com/onmyway133/awesome-voip
- Push Hero - pure Swift native macOS application to test push notifications
- PastePal - Pasteboard, note and shortcut manager
- Frame recorder - Recorder gif and video with frame
- Other apps
- rtpproxy
- Jitter buffer in VoIP
- How to calculate packet size in VoIP
- voice communications
- Voice over IP Overview
- Voice over Internet Protocol
- Open Source VOIP Software
- VOIP call bandwidth
- Routers SIP ALG
- SIP SIMPLE Client SDK
- communications protocol
- Session Initiation Protocol
- RFC 3261
- OpenSIPS - functional, multi-purpose signaling SIP server
- SIP protocol structure through an example
- Relation among Call, Dialog, Transaction & Message
- microSIP
- What is SIP
- What is SIP proxy server
- SIP by Wireshack
- Solving the Firewall/NAT Traversal Issue of SIP
- Introduction to SIP for Java, C#, and VB Developers
- SipML5
- SIP Retransmissions
- draft-ietf-sipping-dialogusage-06
- Creating and sending INVITE and CANCEL SIP text messages
- Kamailio
- Configuring NAT traversal using Kamailio 3.1 and the Rtpproxy server
- How to set up and use SIP Server on Windows
- Build your own VoIP System
- OpenSIPS/Kamailio serving far end nat traversal
- NAT Traversal Module
- RFC
- RFC 3550 - RTP: A Transport Protocol for Real-Time Applications
- RFC 3261 - SIP: Session Initiation Protocol
- Symmetric NAT Traversal using STUN
- RFC 3842 - A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
- Network address translation
- SIP and NAT: Why is it a problem?
- Configuring Port Address Translation (PAT)
- Types Of NAT Explained (Port Restricted NAT, etc)
- One Way Audio SIP Fix
- NAT traversal for the SIP protocol
- A New Method for Symmetric NAT Traversal in UDP and TCP
- SIP NAT Traversal
- NAT and Firewall Traversal with STUN / TURN / ICE
- Introduction to Network Address Translation (NAT) and NAT Traversal
- Transmission Control Protocol
- Datagram socket
- TCP RST packet details
- RST packet sent from application when TCP connection not getting closed properly
- Why will a TCP Server send a FIN immediately after accepting a connection?
- Where do resets come from? (No, the stork does not bring them.)
- TCP listen() Backlog
- Sockets and Ports
- TCP Wake-Up: Reducing Keep-Alive Traffic in Mobile IPv4 and IPsec NAT Traversal
- failed to register using tcp only
- TCP vs UDP
- Transport Layer Security
- Configuring PJSIP with TLS
- Why TLS for SIP
- SIP Signaling Over TLS
- SSL/TLS certificates: What you need to know
- Configuring TLS support in Kamailio 3.1 — Howto
- SIP TLS
- Interactive Connectivity Establishment
- SIP 2012:: ICE — NAT traversal for media
- Introducing pjnath — Open Source ICE, STUN, and TURN for NAT Traversal
- ICE Media Transport
- Session Traversal Utilities for NAT
- STUN
- What is STUN and does it need a port-forwarded server?
- TURN server
- Application Layer Gateway
- What is SIP ALG and why does Gradwell recommend that I turn it off?
- Understanding the SIP ALG
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- About the SIP ALG
- Understanding SIP with Network Address Translation (NAT)
- VoIP — fixing voice quality
- What is Delay in VoIP?
- Understanding Delay in Packet Voice Networks
- Reducing the SIP Packet Size in VoIP
- What Affects Voice Quality in VoIP Calls
- 5 Curable Causes of Poor VoIP Call Quality
- RTP, Jitter and audio quality in VoIP
- An Adaptive Codec Switching Scheme for SIP-based VoIP
- How to master VoIP bandwidth fundamentals
- Voice Over IP — Per Call Bandwidth Consumption
- Echo Analysis for Voice over IP
- Echo Cancellation
- Echo suppression and cancellation
- Echo and Sidetone
- VoIP Echo and how to correct it
- Causes of Echo
- How software echo canceller works?
- DTMF (Dual-Tone Multi-Frequency)
- The dance of DTMF, SIP & RFC 2833 — An introduction
- How can I send inband DTMF tones?
- open source
- PJSUA API — High Level Softphone API
- pjsip library architecture
- pjsip documentation
- Stateful Operations
- Message Creation and Stateless Operations
- Understanding Media Flow
- Getting Started: Building and Using PJSIP and PJMEDIA
- Codec Framework
- Adaptive jitter buffer
- PJSUA-API Accounts Management
- Building Dynamic Link Libraries (DLL/DSO)
- Compile time configuration
- Fast Memory Pool
- SIP and Media Features
- Using SIP TCP Transport
- Monochannel and multichannel audio frame converter
- IOQueue: I/O Event Dispatching with Proactor Pattern
- DNS Asynchronous/Caching Resolution Engine
- Secure socket I/O
- Multi-frequency tone generator
- SIP SRV Server Resolution (RFC 3263 — Locating SIP Servers)
- Exception Handling
- Mutex Locks Order in PJSUA-LIB
- Local Thread Storage
- Thread — Operating System Dependent Functionality
- Threads question
- Using Thread Local Storage
- The Windows Processes and Threads 8
- Example: Thread local storage in a Pthread program
- Thread Local Storage
- Resample Port
- Samples: Using Resample Port
- Memory allocation strategy
- PJSIP — High Performance Open Source SIP Stack
- How to Record Audio with pjsua
- Memory/Buffer-based Capture Port
- File Writer (Recorder)
- Using pjsua to create a mp3 stream
- AMR Audio Encoding
- Audio Device API
- Sound Device Port
- audio bursting
- Buffer problem
- Problem with PJMEDIA’s play callback
- Audio Manipulation Algorithms
- bad quality on iphone 2G with os 3.0
- getting Underflow, buf_cnt=0, will generate 1 frame continuessly
- Measuring Sound Latency - to-end) latency of ​pjsua
- Checking for Network Impairments of Incoming RTP Packets
- Master/sound
- siphon — VIdeoSupport.wiki
- Video Device API - platform video API appropriate for use with VoIP applications and many other types of video streaming applications.
- PJSUA-API Video
- PJSIP Video User’s Guide
- Video streams
- Video source duplicator
- AVI File Player
- PJSIP Version 2.0 Release Notes
- Video API for PJSUA-LIB
- How to make a loopback video call with AVI file?
- What is lib Swscale used for by ffmpeg programers?
- FFmpeg-iOS-build-script
- microSIP - corner/linphone/overview), [Doubango](https://www.doubango.org/), … They all follow strictly SIP standard and may have their own SIP core, for example microSIP uses pjsip, Linphone uses liblinphone, …
- CSipSimple
- user - dev/onmyway133%7Csort:date). Thanks for Regis, I learn a lot about open source and that made me interested in open source.
- What is a branded version
- http://www.gnu.org/licenses/gpl.html
- RTP proxy
- Making RTPproxy work
- Sippy B2BUA and RFC3261 SIP Stack
- rtpproxy(8) — Linux man page
- rtpproxy address filling
- Sample Rate and Bitrate: The Guts of Digital Audio
- VoIP Packet Size
- How to measure brandwith consumption of an rtp stream?
- Sending Reliable Provisional Responses
- SILK audio codec wrapper implementation
- conference bridge questions
- multiple audio devices, multiple calls, conferencing, recording and mix all of the above
- How to Know Offline Call
- Symmetric RTP
- Account specific NAT settings: STUN, ICE, and TURN
- SIP stories, part 3: INVITE retransmission
- TCP/TLS reconnect
- Configuring TCP keep alive and connection lifetime
- Max TCP connections
- VAD detection scenario
- Handle native capture preview
- Video orientation support
- Keep-alive mechanism for TCP and TLS transports
- Periodically transmit RTP packet on silence
- Conference bridge should transmit silence frame when level is zero
- Add user defined NAT hole-punching and keep-alive mechanism to media stream
- Proper way to handle Ip address changes in Android
- support WiFi and 3G simultaneously
- Realtime transport control protocol
- Voice over IP: RTP/RTCP — The transport layer
- RTCP, RTP Control Protocol
- Protocol overview: RTP and RTCP
- The Multimedia Control Protocol RTCP
- Using Conference Bridge
- SIP and RTP Stack
- Media Transport
- RTP Session and Encapsulation (RFC 3550)
- Speex - codec.org/). Also, it’s good to understand the .wav format
- Digital Audio — Creating a WAV (RIFF) file
- Streaming Data from a WAV File
- Programming with Speex (the libspeex API)
- Speex narrowband mode
- Developing with libopus (API reference)
- Building for Other Platforms
- Getting Started: Building for UWP and Windows Phone 8.x
- Porting to New CPU Architecture
- VoIP apps for Windows Phone 8
- How to implement audio streaming for VoIP calls for Windows Phone 8
- In-process, Out-of-process, and Remote Servers
- Basics of an IDL file
- An Introduction to OpenSSL Programming
- SSL Socket support in Windows Phone
- Building Secure Windows Phone 8 Apps — APIs and Techniques
- Is Winsock available on Windows Phone 8?
- Why i am not been able to use all the header files (eg->ssl.h) under open source openssl library
- Getting Started with Winsock
- Building OpenSSL for Visual Studio
- How to Compile OpenSSL for Visual Studio 2010/2012
- OpenSSL for Windows
- OpenSSL for Windows RT
- 2911: enhancement request: Windows RT support
- Need a fast random generator for c++
- Window C/C++ Crypto API Examples and tips
- CryptGenRandom function
- EGD: The Entropy Gathering Daemon
- What does a const pointer-to-pointer mean in C and in C++?
- comp.lang.c Frequently Asked Questions
- What is external linkage and internal linkage?
- Bit Twiddling Hacks
- Better types in C++11 — nullptr, enum classes (strongly typed enumerations) and cstdint
- Microsoft Visual C++ Static and Dynamic Libraries
- Managed C++ — Learn by Example
- Preprocessor directives
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