Speed up testing Kamailio routing

I was very happy to see the news of the release of a new Kamailio module, authored by Victor Sveva.
CFGT can be used to test call scenarios and see what routing logic was triggered in Kamailio.
Test calls need to be marked with a specific, configurable Call-ID pattern ('callid_prefix').
A JSON report is generated, with the possibility to choose what variables to dump into it.
This is going to greatly simplify testing, while potentially keep the logging to a minimum. Highly recommended.

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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