s1p

Rating: 
4.92857
Your rating: None Average: 4.9 (14 votes)

s1p is an experimental SIP client for Sailfish OS

This is software in an early stage of development - please don't expect it to work with every type of SIP server out there.

It also needs a restart after any changes to the settings.

Successfully tested with Asterisk chan_sip.

Please reference the log-id when writing a comment after submittimg a log. The id should be displayed at the bottom of the page if the log was successfully submitted. 

 

Screenshots: 

Keywords:

Changelog: 

- 0.0.2 initial release

- 0.0.3 adds log page

- 0.0.4 fixes registering issues with antisip.com

- 0.0.5 adds ringtones

- 0.0.6 improves UI and makes hanging up calls more reliable

- 0.0.7 fixes some issues with sipgate (inbound calls still broken)

- 0.0.8 fixes inbound calls with sipgate (cancelling outbound calls still broken)

- 0.0.9 fixes issues with outbound calls through sipgate

- 0.1.0 improves UI

- 0.1.1 adds contacts page

- 0.1.2 adds log upload

- 0.1.3 improves logging and handling of audio packets

- 0.1.4 enables audio output selection buttons

- 0.1.5 fixes issues with audio output selection

- 0.1.6 enables mute button

- 0.1.7 adds volume presets

- 0.1.8 makes some changes to how audio frames are handled

- 0.1.9 fixes choppy audio on some phones

- 0.2.0 adds G.711 μ-law codec, (hopefully) fixes some call-state issues

- 0.2.1 adds DNS SRV record lookup

- 0.2.2 fixes call status being sometimes overwritten by regstration status

- 0.2.3 adds initial DTMF support

- 0.2.4 adds preferrence for domain instead of IP in SIP dialogs and adds auth-name field to SIP account settings

- 0.2.5 adds improvements to power consumption, audio handler and playback buffer

- 0.2.6 adds configuration-file options for latency and audio buffer length

- 0.2.7 adds configuration dialog options for latency and audio buffer length

- 0.2.8 fixes issues with setting latency and buffer length

- 0.2.9 enables voicemail button, fixes issues with number input

- 0.3.0 adds options for bind address and regsiter frequency

- 0.3.1 sets default register frequency to 1 hour

- 0.3.2 fixes regsitering with sip.linphone.org

- 0.3.3 fixes previously broken default settings

- 0.3.4 adds support for display name

- 0.3.5 fixes proxy-authentication

- 0.3.6 allows a more flexible approach to call progess messages

- 0.3.7 reinstates stricter approach to call progess messages

Comments

unmaintained's picture

Registering to Antisip works for me in 0.0.4 now. I have no Idea if it is able to make any calls, though.

dubliner's picture

Just a quick note: Registration with Antisip only works when their IP is inserted. Adding "sip.antisip.com" fails (i.e. the phone seems to be trying to connect to itself).
Phone calls are signalled but no audio is transported. Additionally it was impossible to end the call. I had to close s1p.
Anyway, I really enjoy seeing rapid progress here! Keep it up, please!

unmaintained's picture

Audio issues could be NAT related. There's no STUN support in s1p yet. Asterisk seems to just send the audio back to the IP/port combination (when NAT is enabled) the incoming packets are coming from but I don't know how antisip are handling that.
Also the only codec supportet at this moment is alaw but this should not be an issue I think.

dubliner's picture

My SIP devices as well as Asterisk are able to connect to Antisip and other providers through NAT without using STUN. So, there must be another option merely through configuration. I am clearly not an expert, so I'll just try to provide reference:

https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

chan_sip has a setting NAT=yes which never caused any problems.

unmaintained's picture

Yes, you have to use the IP address as it's not resolving domain names at this point. I'll change the description to make this clearer.

unmaintained's picture

Thank you, i'll check the marshaller if it's mayby adding one \n too many or so and chan_sip just doesn't care.

unmaintained's picture

I'm using Asterisk (FreePBX) to test s1p against but I'll try to check out antisip.com tomorrow.

amaretzek's picture

I'm *very* interested! Can contribute with translation to pt_PT, some testing too...

Pages