The developers at Sipwise were very engaged and creative lately, bringing major features in the Kamailio ecosystem:
audio transcoding support in RTPEngine by Richard Fuchs
database API connector implementation for Redis by Andreas Granig (expect a post here about it very soon as well as a presentation at Kamailio World Conference 2018)
Sipwise is one of the oldest companies involved in Kamailio project, since SER/OpenSER times — likely out there in the community are very few that used (or even heard of) the OpenSER Configuration Wizard published by Andreas Granig around years 2006-2007, but that helped many to start building Kamailio-based VoIP platforms back in those days. Andreas, the CTO and one of the founders of Sipwise, has been member of Kamailio management team for more than 10 years now.
Back to the topic of this article, RTPEngine introduced recently the capability of transcoding audio channel for SIP/VoIP calls. It relies on ffmpeg project, therefore the it supports the relevant codecs out there, respectively:
G.711 (a-Law and µ-Law)
AMR (narrowband and wideband)
Another feature added along with the transcoding was the support for repacketization of the RTP traffic, which can help in increasing QoS over long distance connections.
These features are immediately available even on old releases of Kamailio (such as v5.0.x or 5.1.x), the control protocol for RTPEngine being flexible to support such new commands. The commands are not yet documented inside Kamailio’s rtpengine module, but you can read more about them in the README of RTPEngine application:
Along with its old popular feature to gateway between WebRTC DTLS-SRTP and plain RTP (decryption/encryption) as well as the high throughput capacity with in-kernel RTP packets forwarding (useful for NAT traversal or QoS), RTPEngine is nowadays a must-have component in modern Kamailio-based RTC platforms.
Here we express our great appreciation for all these contributions by Sipwise and their continuous support for Kamailio project over the years!