s1p

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s1p is an experimental SIP client for Sailfish OS

Successfully tested with:

  • Asterisk
  • 3CX
  • Yate
  • FritzBox
  • sipgate.de
  • sipgate.co.uk
  • sipnet.ru
  • easyfone.de
  • linphone.org
  • cellip.com
  • easybell.de
  • eventphone.de
  • nfon.com
  • voip.ms
  • voipraider.com
  • peoplefone.de
  • messagenet.com

(please add a comment if you have been able to make it work with a provider not listed above)

Please don't forget to restart the app after making any changes to the settings.

 

Screenshots: 

Keywords:

Application versions: 
AttachmentSizeDate
File harbour-s1p-0.7.9-1.i486.rpm1.85 MB08/10/2020 - 17:24
File harbour-s1p-0.7.9-1.armv7hl.rpm1.82 MB08/10/2020 - 17:24
File harbour-s1p-0.8.0-1.i486.rpm1.85 MB08/10/2020 - 22:46
File harbour-s1p-0.8.0-1.armv7hl.rpm1.82 MB08/10/2020 - 22:46
File harbour-s1p-0.8.1-1.i486.rpm1.85 MB13/10/2020 - 11:40
File harbour-s1p-0.8.1-1.armv7hl.rpm1.82 MB13/10/2020 - 11:40
File harbour-s1p-0.8.2-1.i486.rpm1.86 MB14/10/2020 - 00:37
File harbour-s1p-0.8.2-1.armv7hl.rpm1.82 MB14/10/2020 - 00:37
File harbour-s1p-0.8.3-1.i486.rpm1.86 MB17/10/2020 - 11:16
File harbour-s1p-0.8.3-1.armv7hl.rpm1.82 MB17/10/2020 - 11:16
File harbour-s1p-0.8.4-1.i486.rpm1.9 MB21/10/2020 - 23:00
File harbour-s1p-0.8.4-1.armv7hl.rpm1.87 MB21/10/2020 - 23:00
File harbour-s1p-0.8.5-1.i486.rpm1.9 MB22/10/2020 - 09:43
File harbour-s1p-0.8.5-1.armv7hl.rpm1.87 MB22/10/2020 - 09:43
File harbour-s1p-0.8.6-1.i486.rpm1.9 MB25/10/2020 - 17:08
File harbour-s1p-0.8.6-1.armv7hl.rpm1.87 MB25/10/2020 - 17:08
File harbour-s1p-0.8.7-1.i486.rpm1.9 MB27/10/2020 - 23:32
File harbour-s1p-0.8.7-1.armv7hl.rpm1.87 MB27/10/2020 - 23:32
File harbour-s1p-0.8.8-1.i486.rpm1.91 MB31/10/2020 - 13:41
File harbour-s1p-0.8.8-1.armv7hl.rpm1.87 MB31/10/2020 - 13:41
File harbour-s1p-0.9.0-1.i486.rpm1.91 MB14/11/2020 - 23:12
File harbour-s1p-0.9.0-1.armv7hl.rpm1.88 MB14/11/2020 - 23:12
File harbour-s1p-0.9.1-1.i486.rpm1.92 MB24/11/2020 - 18:57
File harbour-s1p-0.9.1-1.armv7hl.rpm1.88 MB24/11/2020 - 18:57
File harbour-s1p-0.9.2-1.i486.rpm1.92 MB27/11/2020 - 13:14
File harbour-s1p-0.9.2-1.armv7hl.rpm1.88 MB27/11/2020 - 13:14
Changelog: 

- 0.9.2 fixes some issues with FritzBox
- 0.9.1 fixes some issues with linphone.org
- 0.9.0 fixes default primary account setting
- 0.8.9 adds support for multiple active accounts
- 0.8.8 adds auto-answer
- 0.8.7 opens active account section (instead of always the first section) in settings dialog.
- 0.8.6 adds avatar image for ongoing calls
- 0.8.5 fixes disappearing hangup cover action after a call has been answered
- 0.8.4 fixes double entries in call history, fixes lookup of contacts in call history
- 0.8.3 improves audio routing, removes annoying switch to pre-call audio state with last samples still being played.
- 0.8.2 adds log upload to issues tracker
- 0.8.1 introduces (optional) log file (~/.local/share/harbour-s1p/s1p.log)
- 0.8.0 fixes issues with additional incoming calls during an already establiched call
- 0.7.9 introduces changes to the binary size for faster startup
- 0.7.8 fixes deleting call history, adds remorse timer to delete
- 0.7.7 fixes adding SIM calls to history when disabled, adds experimental issues tracker.
- 0.7.6 adds cellular call history integration
- 0.7.5 adds better notification handling, bringing UI to foreground on incoming calls
- 0.7.4 fixes double notifications
- 0.7.3 adds notifications, changes the way SIP daemon and UI communicate enabling background operations in the future.
- 0.7.2 fixes microphone input with headsets
- 0.7.1 fixes issue with receiving rtp traffic when local IP changes
- 0.7.0 changes to external IP address handling, crash handling, length of call-id and tags
- 0.6.9 fixes screen unlock on incoming calls
- 0.6.8 adds support for hardware/headset buttons to configuration page
- 0.6.7 fixes server port data type
- 0.6.6 adds support for additional SIP accounts (only one at a time can be active)
- 0.6.5 testing FritzBox 7590 compatibility
- 0.6.4 adds less used DTMF digits (A-D,F) to pulley menu, reduces ambiguity with status messages
- 0.6.3 minor visual changes to the cover page
- 0.6.2 adds contact name lookup by phone number
- 0.6.1 adds minor visual improvements to the call history and contacts pages
- 0.6.0 adds minor visual improvements to the contacts page
- 0.5.9 adds voicemail icon and counter
- 0.5.8 adds default audio port selector to settings dialog
- 0.5.7 adds Yate compatibility
- 0.5.6 adds audible ringback tone
- 0.5.5 adds compatibility with Easybell
- 0.5.4 testing compatibility with Easybell
- 0.5.3 fixes issues with saving display names in call history
- 0.5.2 adds display name to call history
- 0.5.1 fixes phone number in history page
- 0.5.0 adds call history page
- 0.4.9 adds small visual improvements to the UI
- 0.4.8 improves compatibility with pjsip
- 0.4.7 fixes error in media description parser
- 0.4.6 improves compatibility with 3CX
- 0.4.5 adds rport option
- 0.4.4 allows to set bind port on the settings page
- 0.4.3 adds support for buttons on wired headsets to allow answering / hanging-up calls
- 0.4.2 adds display activation on incomming calls
- 0.4.1 fixes misleadling log entries related to RTP destination address
- 0.4.0 adds codec selector in settings dialog
- 0.3.9 adds locking down to one of the available codecs when answering a call
- 0.3.8 adds workaround for 3cx
- 0.3.7 reinstates stricter approach to call progess messages
- 0.3.6 allows a more flexible approach to call progess messages
- 0.3.5 fixes proxy-authentication
- 0.3.4 adds support for display name
- 0.3.3 fixes previously broken default settings
- 0.3.2 fixes regsitering with sip.linphone.org
- 0.3.1 sets default register frequency to 1 hour
- 0.3.0 adds options for bind address and regsiter frequency
- 0.2.9 enables voicemail button, fixes issues with number input
- 0.2.8 fixes issues with setting latency and buffer length
- 0.2.7 adds configuration dialog options for latency and audio buffer length
- 0.2.6 adds configuration-file options for latency and audio buffer length
- 0.2.5 adds improvements to power consumption, audio handler and playback buffer
- 0.2.4 adds preferrence for domain instead of IP in SIP dialogs and adds auth-name field to SIP account settings
- 0.2.3 adds initial DTMF support
- 0.2.2 fixes call status being sometimes overwritten by regstration status
- 0.2.1 adds DNS SRV record lookup
- 0.2.0 adds G.711 μ-law codec, (hopefully) fixes some call-state issues
- 0.1.9 fixes choppy audio on some phones
- 0.1.8 makes some changes to how audio frames are handled
- 0.1.7 adds volume presets
- 0.1.6 enables mute button
- 0.1.5 fixes issues with audio output selection
- 0.1.4 enables audio output selection buttons
- 0.1.3 improves logging and handling of audio packets
- 0.1.2 adds log upload
- 0.1.1 adds contacts page
- 0.1.0 improves UI
- 0.0.9 fixes issues with outbound calls through sipgate
- 0.0.8 fixes inbound calls with sipgate (cancelling outbound calls still broken)
- 0.0.7 fixes some issues with sipgate (inbound calls still broken)
- 0.0.6 improves UI and makes hanging up calls more reliable
- 0.0.5 adds ringtones
- 0.0.4 fixes registering issues with antisip.com
- 0.0.3 adds log page
- 0.0.2 initial release

Comments

Seven_of_nine's picture

Installed s1p on my wifes and my SFOS Xperia 10's and made two Linphone.org accounts and got back info from linphone.org for both accounts:

SIP identity: sip:username@sip.linphone.org

Mail: my/hermailadress...

Username + Password,

Domain/Proxy: sip.linphone.org

Enterd this into s1p configuration, and left Authentification User +Server Port empty. The other fields like Register Interval (3600), Bind Address (0.0.0.0), Bind Port (5060) I left as it was and made no change.

After restart of program it connects to Linphone.org and reports 'registered'. I have a number keyboard on the screen. On dialing 'normal' phone numbers, some speaker/headphone… icons appear, but there's no connection, and I cannot disconnect, have to close the app and reopen it.

What number has my wifes and my account now? How can I make a Linphone to Linphone call, using my wifes/my usernames? Can only enter numbers. What number have I now on Linphone?

Linphone Homepage is not helpful, therefore I ask here...

 

unmaintained's picture

Which country are you from? Maybe there are easier/better services available than linphone.

unmaintained's picture

There was an authentication issue with linphone.org which has been fixed in 0.9.1
It may still fail occasionally for unknown reasons, though.

carlosgonz's picture

Yes, source code or @unmaintained do not install open code apps in your sailfish os phone. Closed source apps in Openrepos not make sense.

unmaintained's picture

So you mean it would be better not to have any SIP client on SFOS at all?

olf's picture

;-)
@carlosgonz, this has already been discussed in detail here.

> Closed source apps in Openrepos not make sense.
Oh, can you please explain why you think so.

black_sheep_dev's picture

Could you please provide source code?

unmaintained's picture

As discussed somewhere further down the thread it may take a little while until I can release any sources.

 

 

 

olf's picture

@black_sheep_dev, see this thread for details.

emchella's picture
lkdhf's picture

Could anyone give a basic example of how to get it to work with FritzBox? I don't know what I need to enter where.

mklick's picture

Thanks for your app and the ongoing effort. On my Xperia XA and a Fritzbox 7430 is was working some weekd ago. With some audio delay, but it was ok. Now with Sailfish 3.4 and the new FritzOS 7.21 it does not work at all.
Registering takes a long time, then comes the message 'failed' but i am registered. I can be called, but cannot call - the app seems to crash and I cannot hang up. If there is an connection, audio does work not at all, in both directions :-(
I do not know the trigger, if you reactivate the reporting, I want to offer to send logs.

unmaintained's picture

I've re-implementen log uploading into s1p. Please see issue tracker for details. It's a plain text file this time so you can delete/edit any personal information but after upload it's only accessible to admins anyway.

olf's picture

By unmaintained on Sun, 2020/10/11 - 10:36
> Please feel free to open an issue should this problem persist after restarting your phone.

and

By unmaintained on Fri, 2020/10/16 - 22:42
> [...] Please see issue tracker for details. [...]

@unmaintained, please denote in s1p's description above, how to use its issue tracker (e.g., searching for existing issues to check if an issue already has been filed, extending extant ones by adding new info, and filing a new issue).  Is it within s1p?

unmaintained's picture

Yes, I've meant the built-in issues tracker. There aren't many issues reported yet so searching involves scolling down the page but it will be extended as needed once there are too many to be able to confortably browse through.

unmaintained's picture

Please feel free to open an issue should this problem persist after restarting your phone.

unmaintained's picture

The way the UI and daemon communicate changed, that's the main reason there's no logging visible any more inside the app. You could try to start it from the command line to see why it crashes.

zash1958's picture

Works also within an peoplefone SIP-Trunk

www.peoplefone.de / sips.peoplefone.de

PatsJolla1's picture

Works also with Voipraider.com (Login). Can you also integrate how much money is still on the account? And maybe also the server contacts? I have there other contacts then in my phone and don't want to mix them.

unmaintained's picture

Unfortunately I'm not aware of any SIP compliant way to pull your balance or contacts from voipraider.com

oenone's picture

Works with voip.ms

explit's picture

Coul you please build i486 version? Thanks

unmaintained's picture

Could you check if the i486 version is working? I don't have any i486 Saiflish device so I'm flying blind on this one.

michl's picture

S1P seems to work on an FXtec Pro¹ in combination with a FritzBox! Nice work!

A few comments:

- After ending a call and the application the volume of the phone is lower than it was before calling. Using the volume rocker button and raising the volume to the max doesn't restore the original volume setting.

- It seems that the approximation sonsor is not used yet. Is it used on other phones? This might be a special case for the Pro¹.

- Sound quality on both ends isn't really good. It sounds quite muffled.

However, it is quite impressive, that it's working so well! Thanks for your work!

 

unmaintained's picture

Thank you. I don't own a FXtec Pro¹ myself so fixing the issues may involve a lot of guesswork on my part.

The sound seems to be OK on other phones but maybe tweaking the audio settings could have an efefect here (Gain presets etc. on the settings page)

The volume decrease is interesting as there's no code in s1p yet that would change volume settings on SFOS so I'm not really sure why this is happening on the FXtec.

Also the proximity sensor is not used by s1p yet.

unmaintained's picture

Thank you. I will look into giving the available qml and python sources an open-source compliant license and will also explore the best options to add an issues tracker, wiki, etc. as soon as I find some time.

 

andy's picture

Hi unmantained, thanks for the app. I have a fritzbox 7590 and I'm able to receive and make calls perfectly. The only thing that I don't understand is why the status message is "call ended" instead of "registered". I also ask you if would be possible to insert the "R" button in the dialer, as it's necessary for the fritzbox in order to transfer the call to other numbers. Thank you!

unmaintained's picture

According to RFC 2833 DTMF can have the following digits.

Event  encoding (decimal)
0--9                0--9
*                     10
#                     11
A--D              12--15
Flash                 16

 

Does the "R" key work with other SIP phones or would the FritzBox maybe expect a SIP transfer instead? 
Nevertheless, I've added DTMF A-D and Flash to the pulley menu, maybe one of these will do the trick as well.

As for the "call ended" message, it could be that the registering initally fails for some odd reason. You could submit a log and reference the log ID here should this continue to happen.

andy's picture

Unmantained, thank you for all the effort you are putting into this project. I tested the latest version of the app and found the following:
- I confirm that I can make and receive calls (Fritz!box 7590), with a higher quality than the one I get using the Android Linphone or Zoiper apps on Sailfish;
- the status messages are actually much clearer. However, I do not get the message "registered" but "registering failed", and this despite the calls working fine. I have tried to check the log and it actually looks like something is wrong, but I am unable to tell what. If you want to take a look, I sent you the log: id-54. Only once the message "registered" appeared but I was unable to replicate this situation or understand what it depends on;
- I saw that you added the new DTMF digits but I don't think they work. When are they selected should they emit a tone? Because if it has to be, nothing happens. Moreover, not even the dialer digits emit any tone, with the consequence that it is not possible to interact with the automatic responders. Honestly, I don't remember if that was also the case in the previous version. On Linphone when I press a key I hear its tone, I guess it should be the same on s1p too.
Bye!

unmaintained's picture

Thank you for uploading the log. I've tweaked the registration part a litttle to (hopefully) make it more compatible with your FritzBox.

The default method how DTMF is handled with most SIP phones is out-of-band, as described in RFC 4733. This means the events are not transported as audible tones, it's therefore totally normal if you don't hear anything. If the FB is compatible with RFC 4733 it should work nevertheless.
Some other phones may play "convenience" tones locally to indicate a DTMF event has been transmitted but this should not affect the transmission itself.

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