> Do video calls have lower latency audio (assuming the codec can do better than 20ms in the first place), given that there's probably much more data available to send at any given time?
At least for WebRTC, no. The audio and video streams are separate, there's no mechanism in WebRTC to piggyback audio onto video packets. If the receiver is synchronizing audio with video, there's potential for additional delay (but when A/V sync works, it's probably worth it)
I don't know details about WhatsApp calling (although I worked there, I didn't touch the realtime calling stack), I think it does use RTP which isn't really built around piggybacking, but since it's a closed service, they can do whatever. No idea about Apple's calling either.
At least for WebRTC, no. The audio and video streams are separate, there's no mechanism in WebRTC to piggyback audio onto video packets. If the receiver is synchronizing audio with video, there's potential for additional delay (but when A/V sync works, it's probably worth it)
I don't know details about WhatsApp calling (although I worked there, I didn't touch the realtime calling stack), I think it does use RTP which isn't really built around piggybacking, but since it's a closed service, they can do whatever. No idea about Apple's calling either.