Issue #304
If you do VoIP applications, especially with open sources like pjsip, you may encounter kamalio and rtpproxy to serve SIP requests. Due to limitation of NAT traversals, rtpproxy is needed to work around NAT. All SIP handshake requests go …
Issue #284
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 …
Issue #157
This post was from long time ago when I did pjsip
A jitter buffer temporarily stores arriving packets in order to minimize delay variations. If packets arrive too late then they are discarded. A jitter buffer may be mis-configured and be …
Issue #155
As you have probably observed in your studies, there is a determined method for calculating VoIP packet sizes. The packet size depends on many different variables, so there is no great answer for an “average” packet size …
