Continuous Integration and Kamailio

I've presented a workshop at Kamailio World 2016. It focused on tools to help automating the build, deployment and test of Kamailio-based applications using Jenkins, Docker and a few other technologies.

It's been also an opportunity to show a sample usage of the new http_async_client module, designed to perform non-blocking HTTP queries from Kamailio.

The interested reader can find the slides here:

Continuous Integration and Kamailio from Giacomo Vacca

And if you have an hour to spare, here's the full video:

Any feedback or question you may have, please get in touch. I have a post on the event in progress, but there are so many things to highlight that it will require some more time.

Many thanks to Daniel and Elena-Ramona (more info here), event hosts, and Pascom.net for video streaming, recording and editing.

Troubleshooting TURN

WebRTC applications use the ICE negotiation to discovery the best way to communicate with a remote party. I t dynamically finds a pair of candidates (IP address, port and transport, also known as “transport address”) suitable for exchanging media and data. The most important aspect of this is “dynamically”: a local and a remote transport address are found based on the network conditions at the time of establishing a session. For example, a WebRTC client that normally uses a server reflexive transport address to communicate with an SFU. when running inside the home office, may use a relay transport address over TCP when running inside an office network which limits remote UDP targets. The same configuration (defined as “iceServers” when creating an RTCPeerConnection will work in both cases, producing different outcomes.

VoIP calls encoded with SILK: from RTP to WAV

SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates and frame sizes), from narrowband (8 KHz) to wideband (24 KHz). Since Wireshark doesn't allow you to decode an RTP stream carrying SILK frames, I was curious to find a programmatic way to do it. In fact, this has also allowed to me to earn a "tumbleweed" badge in stackoverflow . You may argue that a Wireshark plugin would be the right solution, but that's probably for another day. Initially I thought it was sufficient to read the specification for RTP payload when using SILK ; the truth is that I had to reverse engineer a solution by looking at SILK SDK's test vectors. There, I discovered that a file containing SILK audio doesn't have the file header indicated in the IETF draft ("!#SILK"), but a slightly different one ("!#SILK_V3"). More importantly, each encoded frame is not preced...

Decrypt SDES SRTP from pcap

If you have a pcap file with encrypted RTP (SDES SRTP) and have access to the SIP signalling to see the keys, these instructions will help you decrypt the RTP payload and save it as raw audio. Optionally, depending on the codec, you can then import the raw audio in Wireshark and save it as an audio file. Steps

Giacomo Vacca

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