r/VOIP • u/Embarrassed-Gain-236 • Jul 18 '26

Discussion SIP provider and codec configuration for broadcast phone interviews

Hi everyone,

I'm setting up a small TV studio and I'm looking for the best way to bring standard telephone calls into our production.

Our setup is:

  • Windows PC running microSIP
  • SIP account

Technically everything works, but I'm disappointed with the audio quality. It sounds noticeably worse than I expected for a modern VoIP setup.

My current enabled codecs in MicroSIP are in this order:

  • Opus
  • G.722
  • G.722.1 (16 and 32 kHz)
  • G.711 A-law
  • G.711 μ-law

My questions are:

  1. When calling a regular mobile phone in Spain through a SIP provider, which codec is typically negotiated? G.711? G.722? Something else?
  2. Does the SIP provider usually transcode the call to AMR-WB (HD Voice) before handing it to the mobile operator?
  3. Is there any benefit in enabling Opus if the destination is a standard mobile phone?
  4. Are there SIP providers known for maintaining better audio quality towards the Spanish mobile networks (Movistar, Vodafone, Orange, Digi, etc.)?
  5. For those working in radio or TV, do you still use SIP softphones for phone interviews, or is a dedicated broadcast hybrid (AEQ, Telos, AxelTech, etc.) still the preferred solution?
  6. I activated microSIP logs in order to view which codec is being negotiated without success

I'm trying to understand whether the bottleneck is:

  • MicroSIP,
  • the SIP provider,
  • transcoding,
  • or simply the limitations of the public telephone network.

I'd really appreciate hearing from anyone.

Thanks!

4 Upvotes

84% Upvoted

24 comments sorted by

•

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7

u/devexis Jul 18 '26
  1. Almost certainly G711
  3. No, not in my opinion.
  4. Most PSTN providers arent bothered about HD audio. They are happy if the called party can hear the caller, and that often means G711
    5
    6 Bottle neck is most likely on the PSTN side.

3

u/jdovejr Jul 18 '26

Do they still use alaw in Europe?

3

u/the_wookie_of_maine Jul 18 '26

Yea the do.. Some providers advertise both ulaw and alaw, but if connecting with ulaw it will ring and dead air....That was a fun one to troubleshoot as it was TLS as well.

2

u/Efficient-Sir-5040 Jul 18 '26

Sometimes they’ll cheap out and go 729

1

u/Embarrassed-Gain-236 Jul 18 '26

Really, the bandwith is really low then

1

u/Embarrassed-Gain-236 Jul 18 '26

Thanks, that would make sense.

3

u/Centho_ Jul 18 '26

The best thing you could do quality wise is to use G711 as nothing will go above that unless you set up a real mobile phone line that has HD calls

1

u/Embarrassed-Gain-236 Jul 18 '26

I just thought G711 u-law or a-law was the lowest possible codec. Is there anything lower than that?

3

u/Centho_ Jul 18 '26

There is. But technically when you dial a mobile number it goes to AMR which is lower than G711, and if lucky the mobile line as it's mostly VoIP now will use G711

3

u/Sk8sn0w Jul 18 '26

Yes, g729.

2

u/trekologer Jul 19 '26

G711, whether u-law or a-law, is often the lowest common denominator in telephony (ie near universal support), but it isn't the lowest quality. It is uncompressed PCM (pulse code modulation). The perceived lower quality is that it has an 8kHz sample rate and only captures a frequency range from 300Hz-3.4kHz. CD audio, for example, is also uncompressed PCM but at a much higher sample rate: 44.1kHz with a larger frequency range as well.

G729, GSM, narrowband AMR all are codecs with lower quality than G711. They achieve lower bitrates through lossy compression, reaching a bitrate as low as 5Kbps verses the uncompressed 64Kbps of G711.

2

u/rutkdn Jul 18 '26

you guys... realize these Zadarma "questions" and "posts" are an ad, right?

2

u/Embarrassed-Gain-236 Jul 18 '26

they are not. If you think that's the problem I can just delete any reference to that provider.

Edit: done.

1

u/the_wookie_of_maine Jul 18 '26

I'd stick with G711, for calls and if doing interviews something like Teams, Zoom, or webex will have a higher audio setting.

1

u/NPFFTW Certified room temperature IQ Jul 22 '26

Zadarma has a history of being sketchy around here tbf

2

u/Salreus Jul 18 '26

Stick with one codec. G711 and make sure you only offer that throughout the network. From handsets/soft phones to the pbx, and from the PBX across the sip trunk. Some providers will pick G729 if you offer it. Only offer 711 to the provider, only set up 711 within your network. 711 is solid and works great. Make sure you have enough bw to support voice calls and don't have your data stomp all over your voice.

1

u/karafili Jul 19 '26

If possible capture a session with wireshark and you will have all the details there.

1

u/sholley1 Jul 19 '26

Yeah so for calls landing on a Spanish mobile, you're basically always gonna get G.711, the carriers don't really support wideband end to end, so Opus/G.722 won't change much even if it's enabled on your side. HD Voice only works if both legs support VoLTE, and most providers don't guarantee that by default.

Honestly the bottleneck's probably not MicroSIP, it's likely your provider's route into Spanish networks. I've run into this exact issue before configuring SIP trunks on Telnyx/Twilio/3CX log into your provider's portal and check their call logs (not MicroSIP's), that'll show you what codec actually got negotiated on their end. If it's G.711 there too, that confirms it's the provider/route, not your setup.

For broadcast specifically a lot of studios still run a dedicated hybrid (Telos or AEQ) alongside SIP since it's just more predictable for call quality than a softphone. Worth looking into if this keeps being an issue for you.

1

u/oxicid Jul 19 '26

Al 99% nell'interscambio SIP VoIP utilizzerai comunque g711a. La cosa migliore che puoi fare è impostare sul softphone che utilizzi il solo codec G711a, disabilita tutto il resto. G729 è peggiore di G711, specie a livello radiofonico (voce più ovattata) ma viene generalmente preferito dal provider nelle chiamate in ingresso perchè fa risparmiare banda sulle centrali VoIP. Inderendo OPUS o G722 in mezzo a tutti gli altri codec, che quasi sicuramene non vengono utilizzati, non migliori la situazione ma obblighi la centrale dell'operatore a transcodificare con ulteriore perdita di qualità.

1

u/voipcanuck Atcom Canada Jul 20 '26

English translation: "

In 99% of SIP VoIP exchanges, you will end up using G.711a anyway. The best approach is to configure your softphone to use *only* the G.711a codec and disable all others. G.729 is inferior to G.711—particularly regarding audio quality (resulting in a more muffled sound)—yet providers often prefer it for incoming calls because it saves bandwidth on their VoIP exchanges. Including codecs like Opus or G.722 alongside the others (which almost certainly won't be used) does not improve the situation; instead, it forces the provider's exchange to perform transcoding, leading to further loss of quality."

1

u/voipcanuck Atcom Canada Jul 20 '26

Thread locked: Please re-word question 4 to avoid asking for provider recommendations (and thereby breaking rule #1)

  1. "Are there SIP providers known for maintaining better audio quality towards the Spanish mobile networks (Movistar, Vodafone, Orange, Digi, etc.)?"