awesome-voip
🤙Learning VoIP, RTP, pjsip and SIP
https://github.com/onmyway133/awesome-voip
Last synced: about 24 hours ago
JSON representation
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First of all
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SIP
- OpenSIPS - functional, multi-purpose signaling SIP server
- communications protocol
- Session Initiation Protocol
- RFC 3261
- What is SIP
- SIP protocol structure through an example
- Relation among Call, Dialog, Transaction & Message
- 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
- SIP Retransmissions
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STUN and TURN
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pjsip
- PJSUA API — High Level Softphone API
- open source
- 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
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Audio
- Measuring Sound Latency - to-end) latency of ​pjsua
- 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
- Checking for Network Impairments of Incoming RTP Packets
- Master/sound
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Video
- 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
- 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?
- PJSIP Video User’s Guide
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Codec
- Speex - codec.org/). Also, it’s good to understand the .wav format
- Streaming Data from a WAV File
- Speex narrowband mode
- Developing with libopus (API reference)
- Digital Audio — Creating a WAV (RIFF) file
- Programming with Speex (the libspeex API)
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ALG
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- 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)
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
- What Is Sip ALG (Application Layer Gateway) Voip firewall
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Uncategorized
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VoIP overview
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SIP server
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RFC
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Dual Tone
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NAT
- Network address translation
- SIP and NAT: Why is it a problem?
- Configuring Port Address Translation (PAT)
- Types Of NAT Explained (Port Restricted NAT, etc)
- 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
- NAT and Firewall Traversal with STUN / TURN / ICE
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TCP
- 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
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TLS
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ICE
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Voice quality
- VoIP — fixing voice quality
- What is Delay in VoIP?
- Understanding Delay in Packet Voice Networks
- Reducing the SIP Packet Size in VoIP
- 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
- What Affects Voice Quality in VoIP Calls
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Echo
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Threading
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Resampling
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Memory and Performance
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Porting OpenSSL to Windows Phone 8
- Building OpenSSL for Visual Studio
- An Introduction to OpenSSL Programming
- SSL Socket support in Windows Phone
- 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
- 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
- EGD: The Entropy Gathering Daemon
- Building Secure Windows Phone 8 Apps — APIs and Techniques
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CSipSimple
- CSipSimple
- 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, …
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RTP Proxy
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Other related articles
- 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
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IP change
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RTP and RTCP
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Building pjsip for Windows Phone 8
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C and C++
- What does a const pointer-to-pointer mean in C and in C++?
- 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
Programming Languages
Categories
pjsip
28
Other related articles
21
ALG
17
SIP
17
Audio
16
Porting OpenSSL to Windows Phone 8
13
TCP
12
Voice quality
11
Video
11
NAT
10
RTP and RTCP
9
Echo
7
SIP server
7
TLS
7
Threading
7
Building pjsip for Windows Phone 8
7
C and C++
7
VoIP overview
7
Codec
6
RFC
5
RTP Proxy
4
STUN and TURN
4
Uncategorized
4
ICE
4
Dual Tone
3
First of all
3
Resampling
3
CSipSimple
2
IP change
2
Memory and Performance
2
Sub Categories
Keywords