I think this is likely incorrect based on how much voice/audio distribution meta does today with facebook (and facebook live), instagram and whatsapp - moreso with whatsapp voice message and calling given it's considerable market share in countries with intermittent and low-reliability network connectivity. The fact it is more packet-loss robust and jitter-robust means that you can rely on protocols that have less error correction, segmenting and receive-reply overhead as well.
I don't think it's unreasonable to assume this could reduce their total audio-sourced bandwidth consumption by a considerable amount while maintaining/improving reliability and perceived "quality".
Looking at wireshark review of whatsapp on an active call there was around 380 UDP packets sent from source to recipient during a 1 minute call, and a handful of TCP packets to whatsapp's servers. That would yield a transmission overhead of about 2.2kbps.
quick edit to clarify why this is: you can see starting ptime (audio size per packet) set to 20ms here, but maxptime set to 150ms, which the clients can/will use opportunistically to reduce the number of packets being sent taking into consideration the latency between parties and bandwidth available.
I don't think it's unreasonable to assume this could reduce their total audio-sourced bandwidth consumption by a considerable amount while maintaining/improving reliability and perceived "quality".
Looking at wireshark review of whatsapp on an active call there was around 380 UDP packets sent from source to recipient during a 1 minute call, and a handful of TCP packets to whatsapp's servers. That would yield a transmission overhead of about 2.2kbps.
quick edit to clarify why this is: you can see starting ptime (audio size per packet) set to 20ms here, but maxptime set to 150ms, which the clients can/will use opportunistically to reduce the number of packets being sent taking into consideration the latency between parties and bandwidth available.
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