How to compute or validate the network bandwidth when using Speex

This is a quite simple computation or validation of the network bandwidth, but can be tricky if you don't take into account the "quality" parameter used in Speex.
Speex is designed to change bitrate depending on the desired quality, and there are 10 different levels of quality per type (narrowband (8 KHz), wideband (16 KHz), ultra-wideband (32 KHz)).

All you have to do is take the expected encoding bitrate for that quality (e.g. 28 Kbps for wideband at quality 8/10), compute the payload per frame, optionally sum the payload when you have more than 1 frame per packet, add the overhead of IP, UDP and RTP, then recompute the overall bitrate.

Each Speex frame has 20 msec duration.

The payload for Speex wideband at quality 8 (28 kbps) is:

P = (28*10^3 kbps * 20*10^-3 msec/frame) = 70 B/frame

Consider 1 frame per packet and add the overhead from IP, UDP and RTP:

Ptot = 70 B + 40 B = 110 B/packet

Compute the final network bandwidth:

BW = (110 B)*8 / (20 * 10^-3) = 44 Kbps

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.

Extracting RTP streams from network captures

I needed an efficient way to programmatically extract RTP streams from a network capture. In addition I wanted to: save each stream into a separate pcap file. extract SRTP-negotiated keys if present and available in the trace, associating them to the related RTP (or SRTP if the negotiation succeeded) stream. Some caveats: In normal conditions the negotiation of SRTP sessions happens via a secure transport, typically SIP over TLS, so the exchanged crypto information may not be available from a simple network capture. There are ways to extract RTP streams using Wireshark or tcpdump; it’s not necessary to do it programmatically. All this said I wrote a small tool ( https://github.com/giavac/pcap_tool ) that parses a network capture and tries to interpret each packet as either RTP/SRTP or SIP, and does two main things: save each detected RTP/SRTP stream into a dedicated pcap file, which name contains the related SSRC. print a summary of the crypto information exchanged, if available. With ...

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Giacomo Vacca

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