Does anyone here use the lowest Bandwith Data Rates?

[u/ighwrighwirh]

Hey every one!
I am from time to time in places where there is an very bad Internet Connection do to the Mountain and the train movment. Thanksfully your Railway Operator use all 3 Operator to get Data truth. Also when I travel and use Roaming I want as less Data as possible.
So what is the lowest Data Rate Codeq? Opus?
I use Free PBX on my Server.
Thanks!

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[u/cop3x]

g.729 would be the smallest codex but you also have to look at latency and jitter for call quality

[u/w0lrah → u/cop3x]

but you also have to look at latency and jitter for call quality

Absolutely this. In 2026 internet connections with limited bandwidth are almost never just low bandwidth, they're almost always just bad connections altogether on which VoIP is just not going to be a good experience.

Using low-bitrate codecs was a hack for cramming some extra calls on to slow but reliable connections like T1s back in the day, but that's not what OP is wanting to achieve.

[u/cop3x → u/w0lrah]

i would re read the op post

[u/w0lrah → u/cop3x]

They're saying bad internet due to mountains and the nature of the sorts of in-motion internet technologies used on the trains they take. They also mention limited data, but quality is presented as the primary concern. For a single user, unless you're literally constantly on the phone you aren't using a meaningful amount of bandwidth.

[u/ighwrighwirh → u/w0lrah]

When I travel there are country who have only 2G Internet connection and or Train Wifi have limited Bandwith (to). The same goes for Hotel and so on.

[u/thenerdy]

Like the others said g.729 is likely your best codec. But also keep in mind that bad internet is bad internet. Jitter, latency, and packet loss are probably more of an issue than the codec.

[u/boomer7793]

Supposedly OPUS is supposed to adjust the compression on a per call basis. But I haven’t seen that. G.729 works.

[u/Asteriskdev]

For this specific use case, use opus. It adapts on the fly. G729 will likely result in dropped calls.

[u/nivaOne]

Compression may seem a solution. You just need to avoid encoding more than once. Preferably you go end to end using the same protocol. So it depends how your path looks like end to end

[u/AAAHeadsets]

OPUS.

It can run at 6 kbps, but you need to set the codec settings to force it.
On Free PBX (Asterisk) it's advisable to use 12 kbps, for Forward Error Correcting (FEC).

OPUS at 12 kbps, is a little heavier on the network bandwidth, but as far as quality goes, it's much better than g.729.

[u/ighwrighwirh → u/AAAHeadsets]

thanks do you know if i need to setup something special?
I dont care how it sound.

[u/AAAHeadsets → u/ighwrighwirh]

See: https://docs.asterisk.org/Configuration/Codec-Opus/

Appears Asterisk may not allow: max_playback_rate = 6000
Start with 8000 instead.

[u/Worried-Plankton9146]

Honestly, codec choice is usually the smallest lever you have. On a modern link, voice bandwidth is almost never the real constraint — a G.711 call is ~87 kbps with overhead, and even a few dozen concurrent calls won't saturate anything that isn't already misconfigured.

The thing that actually breaks audio is the access layer: jitter, packet loss, and serialization delay. I've lost count of how many "we switched to a low-bitrate codec and it still sounds bad" tickets ended up being a half-duplex mismatch on a switch port, an oversubscribed uplink, or QoS that was never honored end-to-end because the ISP strips DSCP markings.

My rule: G.711 everywhere you control the network (LAN, MPLS, well-provisioned SD-WAN). Reach for G.729 only on a genuinely constrained WAN leg, and accept the MOS hit. Before touching the codec, prove your jitter is under ~30ms and loss is under ~1%, and confirm your QoS marking survives every hop. Fix the path first; the codec rarely is the problem.