Telegram generally handles low-bandwidth scenarios better than Jitsi for teleconferencing, particularly in practical real-world use on consumer devices and variable mobile networks.
Both platforms use adaptive mechanisms to cope with poor connections (dynamic bitrate adjustment, resolution scaling, audio prioritization), but their design priorities and implementation differ significantly:
Telegram (native app video/voice calls, including group calls):
Jitsi Meet (WebRTC-based, browser or app):
| Scenario | Better Platform | Why Telegram Wins / Jitsi Struggles |
|---|---|---|
| 1-on-1 video call, <500 kbps | Telegram | Stays audio-clear longer; video becomes very low-res but rarely freezes completely. |
| Group call (4–10 people), poor upload | Telegram | Less aggressive per-user upload demand; server-side optimization hides weaknesses better. |
| Mobile data (3G/EDGE), variable signal | Telegram | Native app + MTProto transport more resilient to packet loss/reordering than plain WebRTC. |
| Self-hosted Jitsi on strong server, tuned low-res defaults | Jitsi (possible edge) | Can be forced to very low bitrates manually, but requires configuration. |
| Browser-only, no app install allowed | Jitsi (slight edge) | No choice — but expect more complaints on weak links. |
If your typical teleconferencing happens on mobile data, spotty hotel Wi-Fi, or developing-region networks, Telegram is the safer bet for staying intelligible and connected. For privacy-focused or highly customizable group meetings where you can control the server and educate participants to turn off video early, Jitsi can be made to perform acceptably — but it requires more effort.
If you're designing monitoring/logging around this (bandwidth spikes, packet loss patterns, adaptive bitrate decisions), Telegram's proprietary nature makes deep RF/spectral analysis harder, while Jitsi (open WebRTC) lets you capture and dissect negotiation/RTCP reports much more precisely with tools like Wireshark or custom WebRTC stats logging.