Success With SIP Service

For those of you here who are thinking of using a SIP service on your Librem 5, I thought I would share some information here that might make it easier for you. If you are as completely clueless about this process as I was until just last night, then this information will come in very handy for you. I spent several hours learning to complete what should have been a half-hour process because I started knowing nothing. And I like to learn incrementally. So instead of adding the complexity of a Librem 5 and Linux to start, I installed and configured the SIP service on to my Pixil 6 Pro (running Grapheneos) phone. With what I know now, it shouldn’t be too hard to repeat the process on my Librem 5 now. Especially now that I already have an established SIP account.

I started by following an online thread that said to use the Jami app, along with the SIP service. The time I spent signing up with Jami and installing Jami was a complete waste of time, like about an hour before I realized that Jami didn’t have the needed fields to input the right information. It took another two hours after signing up at and paying for my first month, before I was able to find the critical information to even configure a SIP service on to a phone app. So there went three frusterating hours before I started having any success at all. So let me save some of you, a great amount of frusteration here. The steps are listed below.

1.) Go to: and sign up for an account. This part is free. Your email address is your login name and you’ll make up a password. This login password will end up being your SIP login password also.

2.) Login, go to the “Finance” tab at and add a minimum of $15.00 to your account via either Pay Pal or a credit card. I am going to wait to set up recurring billing until after I have used the service for a while first.

3.) Either on the Main Menu at, or on the DID number Menu (I forget which), you order the SIP account that you just paid for. It’ll cost less than $5.00. But the company expects your open balance with them to never drop below $9.00. You’ll fill out several tabs that may barely make sense and might not assure you that you’re doing things correctly. I filled in every field that I understood which was mainly with my personal information and that indicated that I wanted to get a SIP service. A lot of more information auto-filled with technical information after I saved what I did select. After filling in several tabs, the information that I would need later was found amongst a lot of technical information from several fields. There was actually way too much information there. I still didn’t have a clue how to set anything up on my phone. But I did one important thing correctly. I entered my main (current) phone number in to a field as my “Caller ID number”. This is the number that spoofs your current number to tell everyone who you call via SIP, that it is you calling, even though you’re calling from a phone with a different main phone number. will text you a code to that number that you enter, to verify that you own that phone number. Alternately, you can buy a new number from them for a few dollars and make that number be your caller ID number.

4.) Download and install a SIP client app to your phone. Jami didn’t work. But a free version of Zoiper worked well. Later I found two other SIP apps that both worked also. But Zoiper is the one that I’ll use for a while to start with.

5.) Out of everything found in the tabs in my account, only three pieces of information mattered when setting up the account on my phone. 1.) Your “SIP/IAX Main User Name”. This will be a six digit number. 2.) The Server Name, and 3.) Your login password. A valid Zoiper login name that you enter in to the app would look like “”. And then you enter your login password. That’s it. You’ll see the Zoiper app verify verify that the server accepted everything. And then SIP on your phone is ready to use. With other apps, you only need the same information to login. But instead of using the “@” symbol, in some other SIP apps there are separate fields for the user name and server name.

6.) After verifying that SIP calls work and that my SIP caller ID number is correct (my number from the other phone), I added my Pixil 6 Pro phone number in to Google Voice as another number to forward my calls to. So when people call my number that they know me as, I can answer it on my Pixil 6 now (soon to be also on my Librem 5 now). If I call them, I use the SIP app to place the call. So they’ll know it’s me calling before they answer.

7.) After this all works on my Librem 5 also, I plan to port my Google Voice number (that’s my main long-term number) to my Pixil 6 Pro and then ditch Google Voice all together. Then using my Librem 5 as a daily driver will only require forwarding all calls incoming to my Pixil 6 Pro, to the Librem 5. It only takes dialing a few codes in to the Pixil 6 Pro, to forward the calls. Other than myself, no one ever needs to even know the Librem 5’s real phone number. And if an important event comes up where I need a lot of Google apps, I can then still take the Pixil 6 Pro if/when needed, and not miss any calls. But if you’re ever going to Washington DC to protest at the Capital, be sure to bring your Librem 5 with an AwSim in it and don’t forward your calls.

Just another piece of information here. When I add my SIP account information in to Purism’s phone dialer on my Librem 5, I get a confirmation that my SIP service is connected to the phone. But the phone dialer doesn’t work. I thought it should work via wifi. I don’t have a SIM card for this Librem 5 yet. I don’t know if a SIM card is needed tonget SIP working or not.

Another update. I just made my first call from my Librem 5. There is no SIM, just the SIP account. I could hear the other caller. But she couldn’t hear me.


I’ve played with the sip function as i have my own freepbx instance set up on my vps and it does work but also have the one way audio issue. Must be something on the L5 that isn’t fully implemented yet.

1 Like

This sounds like a problem that a number of us had with the default Calls app on the L5, using and some other SIP providers. After a very long thread of trying different possibilities, we narrowed it down to a mismatch of audio codecs.

Since you say you don’t know much about the technical aspects, I won’t ask you to find which codecs the Zoiper app allows you to use. Instead, I’ll suggest the solution that worked with Calls and, changing one of the Account Settings on the Customer Portal. If it works, all to the good. If it doesn’t work, you can revert your settings to the way they were before, and pursue a different line of investigation.

  1. Starting at the Portal: Home Page, mouse over “Main Menu” and click “Account Settings” in the drop-down menu that opens up.

  2. Click on the “Advanced” tab (on the far right).

  3. “Allowed Codecs” is the third line down. Uncheck G.711U

  4. Make sure that G.722 is checked. (It used to be unchecked by default and marked “Experimental”, but it now seems to checked by default.)

  5. Click “Apply” at the far right of the “Allowed Codecs” line.

  6. Try your phone call.

I hope it works.

the VOIP service requires upload of official government ID:

Thank you for choosing We look forward to seeing you get started with our award-winning platform.

As an international VoIP service provider, we are committed to complying with the legal and regulatory frameworks applicable to our company in the jurisdictions where we conduct business, and we are advised to execute diligent security checks for all new accounts including, but not limited to, Know Your Customer (KYC) standards.

Your account has been put on hold while our team runs verification on the information provided. We value your trust in our services, and we are committed to ensuring a secure and efficient verification process for all our customers.

You can also help expedite this procedure by submitting additional documentation in advance such as an official government identification. You may send any additional documentation to our team directly at or by uploading it below.

We understand that sensitive information demands extra care, and we want to assure you that your data will be handled with the utmost confidentiality and in strict accordance with our privacy policy.

Rest assured that once your documents are received, they will be reviewed promptly, and we will keep you informed about the progress of your verification.

If you have any questions or concerns regarding this request or the verification process in general, please do not hesitate to contact our customer support team at or via Live Chat. 

Non-Starter for me. You don’t even have to do that for signing up with Verizon, or Comcast, or AT&T, it seems VOIP providers have very stringent user identification requirements.

Ideally Purism should create it’s own VOIP service and integrate it with the phone app, that would make signup a lot easier.

1 Like

Yes, I got two way voice working on my Librem 5, using SIP on the Gnome dialer program. You do have to set codecs to use only the oldest codec, by turning off the two other codec choices. So you get two way audio then. But it sounds really bad on the other caller’s phone.

I also bought an old fashioned phone handset from Amazon. Not the phone base, just the handset with the older style coiled phone cord. It terminates with a 3.5mm plug. I plugged that jack in to myWindows 10 PC and used a free Windows program called MicroSIP. It works flawlessly the first time. All of the modern codecs are there and voice quality is both directions is excellent. Then I plugged that handset in to my Ubuntu 20.04 PC. Both PC’s are identical NUC6s. The built-in software store in Ubuntu has multiple different SIP programs. I installed and tried all of them. None of them worked. They all had old codecs that were older than the one on my Librem 5.

Here is a link to that handset.

If you are setting up VoIP to use at home, results will likely be better if you bypass the PC and configure a separate ATA (Analog Telephony Adapter). You will still need a PC or other computer to do the configuration, but the phone calls will pass directly from your ATA through your router/modem and out to the internet.

Regarding the choice of codecs, it is not necessarily a question of how old they are, but more about how well they tie in with the telephone system at the other end of your call and how much bandwidth they need. There is likely to be more latency (bad) if the codec needs more bandwidth.

ISP contracts usually provide much slower upstream speeds than the (advertised) combined up+down. If you use a bandwidth hog like G.711, your upstream (i.e. the audio from your voice) could be bumping against its speed limit. That would make for poor listening at the other end.

Most people find latency to be very annoying in a phone call, making it worthwhile to sacrifice a very small amount of speech quality for a considerable reduction in latency. I have had good success with the rather old G.729a for the primary codec. This is not FOSS and is therefore frowned upon by Gnome enthusiasts.

Distributed Linux SIP software tends to be pretty stale, yes.

I can recommend Blink which is an excellent SIP client on Mac, Windows and Linux.

Also a current release of linphone ( works very well

As for SIP on the L5, I successfully used it with the built in SIP support in gnome-calls. But it tends to be a tad fragile, as - depending how you connect - NAT and packet filtering can easily break things.

Thanks for the information.