Hi, Really enjoying using Vapi - its the best service for building voice bots. Although sometimes my calls are of the crazy bad quality. Like its just noize, and the speech cannot be recognized. Its weird because it happens on calls taken in quiet rooms in ideal environments. Is this something on the Twilio side? Any advice on how to fix it, or to debug? I attached the example of the recording downloaded from the Vapi. This call was taken in the ideal environment. I have 0 idea why the quality is so bad. https://cdn.discordapp.com/attachments/1366869224798031953/1366869225028976780/twilio-vapi-user-recording.wav?ex=681283aa&is=6811322a&hm=d05e6f46124dfa01256c0796a5d3fee88e1e4ba0fd721822c0025a216acdc754&

@ihorcodes Hmmm.... Can you provide a slightly longer recording? it's a little hard to hear anything there.

yeah, that's the point. the part when user is talking is unrecognizable. in the attached recording i kept the part when the bot it talking. https://cdn.discordapp.com/attachments/1366869224798031953/1366874782691491866/vapi-twilio.wav?ex=681288d7&is=68113757&hm=397a63af0a98e0033307f032687a65a14e6644d88525e4b55fd20fb40c5c00ad&

Hey! To help track down this issue, could you share: - The call ID This would really help us figure out what went wrong!

Call ID: 679707fb-1a6d-44ce-ae86-29885346f32f

it sounds like its twilio related. the Vapi assistant sounds clear and only the user side is a bit fuzzy. I would run some tests with just twilio just to try and isolate the issue. on your side, is the assistant sounding clear to you?

thanks for confirming this. yes, i bought few numbers with Telnyx and i haven't experienced same issues

glad we could confirm that. let us know if you have any issues with Telnyx or other providers regarding the same issue so we can dive deeper into the issue.

I’m having a similar issue to the one mentioned above, but with Telnyx. The AI sounds perfectly clear on my end, but it can barely hear me. When I check the call logs, my voice is super compressed and almost unintelligible, so it's no surprise the agent can’t understand. Just before that, I had a regular phone call with a friend — no issues at all, they heard me perfectly — so it’s clearly not a problem with my phone. Here’s one example with the audio file attached: Call ID:

b8754efb-ea17-49fb-9df6-5e8a43cb32dc

Same issue on another call: Call ID:

41690194-3dcb-45d3-84ff-933f10c41941

Let me know if you have any idea where this might be coming from. https://cdn.discordapp.com/attachments/1366869224798031953/1369694471368867860/b8754efb-ea17-49fb-9df6-5e8a43cb32dc.mp3?ex=681ccae1&is=681b7961&hm=daf52bd689d4312f054834ed3a33cfbf39492e15e5c00fc3a7fb7eba174ad474&

@Valentin | Noodek.com @ihorcodes Now that I have a bit more context, and the recordings I downloaded and analyzed - here are my comments: - The issue you're exhibiting is not related to VAPI, it is related to the communications provider - there is no question about it. - It doesn't matter if you route it via Twilio or Telnyx, or any other carrier in the world for that matter, it could - and most probably would, happen again. - The audio issue you're experiencing is exhibited due to Network Jitter, causing audio packets to simply get lost in transit, what you hear as gaps in the audio. Now, here are a few things you need to remember: - Telnyx, Twilio, Plivo, or any other comapny has infrastructure in pre-set locations, not in every country in the world. - Calls are routed from the point of origin or to the point of destination, based upon Internet connectivity - and the Internet isn't a bullet proof place. - Carriers normally use g711 based codecs. These are 64kbps consuming, over an 8KHz sample rate. The recording of the VAPI side sounds more like 128kbps over a 16KHz or even a 24KHz sample rate. This means that the carrier must transcode the audio from VAPI, and vice versa - which introduces another latency and audio loss (when you transcode, you always lose some of the data, mathematically speaking). All things considered - when working in "exotic locations" (eg. not any of the following: US, West Europe, Middle East, Pacific) - you may run into these issues, depending on the carrier and their business relationships in the region.

Okay, but as for me, I live in Paris (France, West Europe), I’m with Orange (the best carrier in France), and my phone uses Wi-Fi for calls (so the quality is superior). Moreover, the "SIP Trunking" mode in Telnyx works perfectly, whereas the "Programmable Voice" mode works very, very poorly (as previously mentioned). So I’m having a hard time understanding what you’re telling me.

Well, programmable voice and elastic SIP trunking use 2 very different systems. Elastic SIP trunking is geared for improved transcoding and interconnecting capability

In the case of the "Programmable Voice" mode, what do you make of the first line of my message?

What do you mean ?

I mean that what you're highlighting is at odds with what I'm observing on my end. You're talking about network instability and various other issues, but in my case, pretty much everything is aligned for things to work properly. That's why I don't understand where the problem is coming from. In the end, my issue is more or less resolved, since if I switch to "SIP Trunking" mode, it's perfectly stable. The problem I have with this method is that call forwarding to another number doesn't work (for example, when the caller asks to speak to a human). Whereas call forwarding works just fine in "Programmable Voice" mode... It's never-ending...

Did you enable

SIP REFER

support in your SIP trunk?

I’m not really sure what it is, and I don’t see it anywhere in the Telnyx interface (although I did come across some references to it in their API documentation).

Telnyx doesn't support that feature. However, what you can do is pass your telnyx SIP trunk via cloudonix (or a similar tool) and then by implementing the relevant API via an agent tool, you can obtain a similar functionality or even a better one.

@Nir Simionovich (CEO@Cloudonix) how to do that do i have to pay an additional fee for the transfer call?

@Toketo1 What do you mean, which "fee" are you referring to exactly ?

Telnyx doesn't matter much to me, so what other solution can I use to do something that seems particularly basic to me on Vapi?

@Valentin | Noodek.com One of the fundamental problems with communications and telephony in general, is that it is considered simple and basic - while in fact, it is one of the most complicated pieces of technology and regulations ever invented. I've been in the telecom industry for the past 26 years - let's put it this way, picking up the phone and dialing, and getting the other side to answer - involves hundreds of little systems working together, making things happen. Your other option is to "go solo" - or to be more accurate - "take control of your fate". Personally speaking, I don't like buying numbers and services from Twilio/Telnyx and such, because I know very well they are marking up over 200% on the price (sometimes even 600%). For example, a call minute in the US costs between $0.004 to $0.006 per minute - Twilio/Telnyx sells is for $0.0115. Now, considering their service and what they do, it's a reasonable markup. But in other places, the difference is crazy. For example, in some countries they sell a minute for $0.12 per minute, while the actual price should be $0.012 - that's a 900% markup. So, my point is this - if you are located in France for example, you can get a SIP Trunk from a local carrier, and try to connect it directly to VAPI. If it works - GREAT, you saved alot of money. If it doesn't work directly, you can use Cloudonix to connect it - and you'll still save alot of money on your calls and numbers.

Thank you for your feedback, I understand the point you're making. One thing I perhaps should have clarified from the start is that the call forwarding needs to be directed to a dedicated landline. So, in principle, if it's a VOIP line, could that solve my issue, since I wouldn't need to go through the traditional phone system? Can Vapi handle forwarding to a VOIP SIP line?

Yes and no

I found this post mentioning the availability of "SIP REFER": https://telnyx.com/release-notes/transfer-calls-with-sip-refer-live And there's an example in the Vapi documentation with Telnyx: https://docs.vapi.ai/advanced/sip/sip-trunk#sip-refer-call-transfer

The transfer shown is internal, namely, an inbound call via a Telnyx number to a PSTN number via Telnyx. I'm doubtful this method will work on a non-telnyx URI.

@Toketo1 @Nir Simionovich (CEO@Cloudonix) @Valentin | Noodek.com Do you happen to have the issue that, with SIP integration, Vapi doesn't identify rejected calls and voicemail? When it was integrated with the key, it could identify them. Now, it treats everything as answered and charges for it, even rejected calls.

You mean rejection post transfer?

No, simply when a call is rejected by the user or not answered.

Hi Valentin. How did you set this up in Vapi?