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Learning VoIP, RTP and SIP (aka awesome pjsip)

onmyway133 profile image Khoa Pham Updated on ・23 min read

Learning VoIP, RTP and SIP (aka awesome pjsip)

Before working with Windows Phone and iOS, my life involved researching VoIP. That was to build a C library for voice over IP functionality for a very popular app, and that was how I got started in open source.

The library I was working with were Linphone and pjsip. I learn a lot of UDP and SIP protocol, how to build C library for consumption in iOS, Android and Windows Phone, how challenging it is to support C++ component and thread pool in Windows Phone 8, how to tweak entropy functionality in OpenSSL to make it compile in Windows Phone 8, how hard it was to debug C code with Android NDK. It was time when I needed to open Visual Studio, Xcode and Eclipse IDE at the same time, joined mailing list and followed gmane. Lots of good memories.

Today I find that those bookmarks I made are still available on Safari, so I think I should share here. I need to remove many articles before they are outdated or not available anymore. These are the resources that I actually read and use, not some random links. Hopefully you can find something useful.

This post shows more resources for pjsip on client and how to talk directly and with/without a proxy server.

First of all

Here are some of the articles and open sources made by me regarding VoIP, hope you find it useful

VoIP overview

Voice over Internet Protocol (also voice over IP, VoIP or IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet

  • Voice over IP Overview: introduction to VoIP concepts, H.323 and SIP protocol

  • Voice over Internet Protocol the wikipedia article contains very foundation knowledge

  • Open Source VOIP Software: this is a must read. Lots of foundation articles about client and server functionalities, SIP, TURN, RTP, and many open sources framworks

  • VOIP call bandwidth: a very key factor in VoIP application is bandwidth consumption, it’s good to not going far beyond the accepted limit

  • Routers SIP ALG: this is the most annoying, because there is NAT and many types of NAT, also router with SIP ALG

  • SIP SIMPLE Client SDK: introduction to SIP core library, but it gives an overview of how

SIP

The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, as well as in instant messaging over Internet Protocol (IP) networks.

SIP server

  • Kamailio: this is the server that I used, and it plays well with lots of standard SIP clients, including pjsip. Debugging on this server was also a fun story

RFC

RTP, SIP clients and server need to conform to some predefined protocols to meet standard and to be able to talk with each other. You need to read RFC a lot, besides you need to read some drafts.

NAT

NAT solves the problem with lack of IP, but it causes lots of problem for SIP applications, and for me as well 😂

TCP

Learn how TCP helps SIP in initiating session and to turn in TCP mode for package sending

TLS

Learn about Transport Layer Security and SSL, especially openSSL for how to secure SIP connection. The interesting thing is to read code in pjsip about how it uses openSSL to encrypt messages

ICE

Learn about Interactive Connectivity Establishment, another way to workaround NAT

STUN and TURN

Learn about Session Traversal Utilities for NAT and Traversal Using Relays around NAT, another way to workaround NAT

ALG

Learn about Application Layer Gateway and how it affects your SIP application. This component knows how to deal and modify your SIP message, so it might introduce unexpected behaviours.

Voice quality

Learn about voice quality, bandwidth and fixing delay in audio

Echo

This is a very common problem in VoIP, sometimes we hear voice from the other and also from us. Learn how echo is made, and how to effectively do echo cancellation

Dual Tone

Learn how to generate dual tone to make signal in telecommunication

pjsip

PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile handsets.

Threading

pjsip uses Local Thread Storage which introduces very cool behaviors

Resampling

How to work with sample rate of the media stream

Memory and Performance

Audio

Video

I learn a lot regarding video capture, ffmpeg and color space, especially YUV

CSipSimple

There are many SIP client for mobile and desktop, microSIP, Jitsi, Linphone, Doubango, … They all follow strictly SIP standard and may have their own SIP core, for example microSIP uses pjsip, Linphone uses liblinphone, …

Among that, I learn a lot from the Android client, CSipSimple, which offers very nice interface and have good functionalities. Unfortunately Google Code was closed, so I don’t know if the author has plan to do development on GitHub.

I also participated a lot on the Google forum for user and dev. Thanks for Regis, I learn a lot about open source and that made me interested in open source.

You can read What is a branded version

I don’t make any money from csipsimple at all. It’s a pure opensource and free as in speech project.
I develop it on my free time and just so that it benefit users.
That’s the reason why the project is released under GPL license terms. I advise you to read carefully the license (you’ll learn a lot of things on the spirit of the license and the project) : http://www.gnu.org/licenses/gpl.html
To sump up, the spirit of the GPL is that users should be always allowed to see the source code of the software they use, to use it the way they want and to redistribute it.

RTP Proxy

Because of NAT or in case users want to talk via a proxy, then a RTP proxy is needed. RTPProxy follows standard and works well with Kamailio

Other related articles

IP change

IP change during call can cause problem, such as when user goes from Wifi to 4G mode

RTP and RTCP

Learn about Realtime transport control protocol and how that works with RTP

Codec

To reduce payload size, we need to encode and decode the audio and video package. We usually use Speex and Opus. Also, it’s good to understand the .wav format

Building pjsip for Windows Phone 8

Windows Phone 8 introduces C++ component , changes in threading, VoIP and audio background mode. To do this I need to find another threadpool component and tweak openSSL a bit to make it compile on Windows Phone 8. I lost the source code so can’t upload the code to GitHub 😢. Also many links broke because Nokia was not here any more

Porting OpenSSL to Windows Phone 8

Firstly, learn how to compile, use OpenSSL. How to call it from pjsip, and how to make it compile in Visual Studio for Windows Phone 8. I also learn the important of Winsock, how to port a library. I struggled a lot with porting openSSL to Windows RT, then to Windows Phone 8

A lot of links were broken 😢 so I can’t paste them all here.

C and C++

Since pjsip, rtpproxy and kamailio are all C and C++ code. I needed to have a good understanding about them, especially pointer and memory handling. We also needed to learn about compile flags for debug and release builds, how to use Make, how to make static and dynamic libraries.

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Discussion (7)

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bdmorin profile image
Brian

Wow.. this is an exceptional list. Should make a github awesome list from it.

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martsa profile image
martsa • Edited

Created a simple sip client using native android sip app , but later came to know that it doesnt support all android devices. So thought of developing using PJSIP. If you can suggest me ,where i can get a very simple android native pjsip call accept and send demo source code. i find it little complex to search . simpler than the csip.

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kristiallen939 profile image
Kristi Allen

Very well written! Having read this, i thought its informative content. Thanks for sharing detail information about online phone number/ VOIP, SIP and RTP. callhippo.com/online-phone-number

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ameedjamous profile image
Ameed Jamous

I would add github.com/datso/react-native-pjsip to the list. it's a new react native PJSIP project that looks interesting.

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mf777star profile image
MF777-star

Wow, you know the feeling when you leave a very informative course or lecture? That's how I feel after reading your very comprehensive post here. Thank you! I use VoIP through my business phone systems. Particularly for international toll free numbers I have found VoIP extremely beneficial for my business. The cost savings compared to traditional systems as well as a modification of the old way of thinking that you had to have a brick and mortar location in order to really be present in international markets has been very altering for our business model. Thank you again for your extremely comprehensive and informative post!

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onmyway133 profile image
Khoa Pham Author

Hi mate, glad that you find my post useful.

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