Voip Service API For Legato .... [Linphone]

Greetings folks i have built a working voip stack based on linphone for legato+mangoh … using belle-sip mediastreamer2 and linphone … the idea would be to implement this as a service API not a standalone program …

i have added a legato shell audiofilter [driver] to mediastreamer to use the le_audio bits it fires up 2 threads one for read one fro write and using recorder/player diverts the audio to/from speaker/mic [possibly other streams] …

when done the console phone will be ported to a service api le_voip… if there is any interest or feed back it will be welcomed.

Greg

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Hi Greg, I apologize for not getting back to you earlier. We definitely would like to hear about your voip service. Is the intention to eventually contribute back to the Legato community? Are you planning to commercialize this product?

This is a pure opensource project … the end product is a commercial product that will have a legato abstraction layer. the problem i am having at the moment is that its not possible to read and write from the PCM simultaneously a playback/record is required for VOIP to work … the cli can make / receive calls but cannot process audio due to this limitation.

Greg

Hi Greg,
Why do you say you’re not able to read and write from the PCM simultaneously? It should be possible.

Tristan

Agree it should be possible :smiley:

i am using the WP8 …

"Other audio features and constraints:

DTMF decoding works only on MdmVoice Rx stream. DTMF’s reception handler must be installed before call initiation.
File Playback/Recording: WAVE, AMR Narrowband and AMR Wideband formats are supported.
Samples insertion/extraction: PCM samples using configuration of channel number, sampling rate and sampling resolution.
Several file playbacks can be activated simultaneously but not several file recordings. Playback and recording can’t be performed simultaneously. This also applies to samples insertion/extraction."

Hi Greg,
I’m afraid our documentation is quite outdated. :confused:
Actually, it should be possible on WP85 if you start the playback before the recording.
Please let me know.
Kr,
Tristan

This will be a priority once the first WP board goes to market… In a few weeks

Hi Greg,

It was nice meeting you F2F today. Please give Tristan’s explanation a try and let us know how it goes.

Thanks,
Alan

Been non stop since i got back will do great meeting you and the team

Have you released this as open source yet? I’m very interested in this project.

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I have got a build of it into the rootfs have not tested it much the problem has been audio support does not fully function i am now planing on running it via a USB sound card i am back on this project now.

Blockquote
root@swi-mdm9x15:~# /lib/ld-linux.so.3 --list /usr/bin/linphonec
liblinphone.so.9 => /usr/lib/liblinphone.so.9 (0xb6ed3000)
libbctoolbox.so.1 => /usr/lib/libbctoolbox.so.1 (0x43440000)
libortp.so.13 => /usr/lib/libortp.so.13 (0x432f0000)
libmediastreamer_base.so.10 => /usr/lib/libmediastreamer_base.so.10 (0xb6ebb000)
libc.so.6 => /lib/libc.so.6 (0x43108000)
libbellesip.so.0 => /usr/lib/libbellesip.so.0 (0xb6c1d000)
libmediastreamer_voip.so.10 => /usr/lib/libmediastreamer_voip.so.10 (0xb6bae000)
libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb6a87000)
libbelr.so.1 => /usr/lib/libbelr.so.1 (0xb6a20000)
libbzrtp.so.0 => /usr/lib/libbzrtp.so.0 (0xb6a08000)
libz.so.1 => /lib/libz.so.1 (0xb69ee000)
libsqlite3.so.0 => /usr/lib/libsqlite3.so.0 (0x43578000)
libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x43680000)
libm.so.6 => /lib/libm.so.6 (0x43470000)
libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x43638000)
libpthread.so.0 => /lib/libpthread.so.0 (0x43240000)
librt.so.1 => /lib/librt.so.1 (0x43560000)
libdl.so.2 => /lib/libdl.so.2 (0x432e0000)
libpolarssl.so.7 => /usr/lib/libpolarssl.so.7 (0x43268000)
/lib/ld-linux.so.3 (0x430d8000)
libantlr3c.so.1 => /usr/lib/libantlr3c.so.1 (0xb69d0000)
libsrtp.so.1 => /usr/lib/libsrtp.so.1 (0xb69b3000)
libgsm.so => /usr/lib/libgsm.so (0xb69a4000)
libopus.so.0 => /usr/lib/libopus.so.0 (0xb695b000)
libspeex.so.1 => /usr/lib/libspeex.so.1 (0xb693e000)
libspeexdsp.so.1 => /usr/lib/libspeexdsp.so.1 (0xb6924000)
libasound.so.2 => /usr/lib/libasound.so.2 (0xb6860000)
libavcodec.so.53 => /usr/lib/libavcodec.so.53 (0xb5de5000)
libavutil.so.51 => /usr/lib/libavutil.so.51 (0xb5dc0000)
libswscale.so.2 => /usr/lib/libswscale.so.2 (0xb5d97000)
libtheora.so.0 => /usr/lib/libtheora.so.0 (0xb5d4d000)
libvpx.so.1 => /usr/lib/libvpx.so.1 (0xb5bd0000)
libtheoraenc.so.1 => /usr/lib/libtheoraenc.so.1 (0xb5b93000)
libtheoradec.so.1 => /usr/lib/libtheoradec.so.1 (0xb5b79000)
libogg.so.0 => /usr/lib/libogg.so.0 (0xb5b6c000)

linphonec> help
Commands are:
---------------------------
      help	Print commands help.
    answer	Answer a call
autoanswer	Show/set auto-answer mode
      call	Call a SIP uri or number
     calls	Show all the current calls with their id and status.
 call-logs	Calls history
    camera	Send camera output for current call.
      chat	Chat with a SIP uri
conference	Create and manage an audio conference.
  duration	Print duration in seconds of the last call.
  firewall	Set firewall policy
    friend	Manage friends
      ipv6	Use IPV6
      mute	Mute microphone and suspend voice transmission.
       nat	Set nat address
     pause	pause a call
      play	play a wav file
playbackga	Adjust playback gain.
     proxy	Manage proxies
    record	record to a wav file
    resume	resume a call
 soundcard	Manage soundcards
      stun	Set stun server address
 terminate	Terminate a call
  transfer	Transfer a call to a specified destination.
    unmute	Unmute microphone and resume voice transmission.
    webcam	Manage webcams
      quit	Exit linphonec
---------------------------
Type 'help <command>' for more details or
     'help advanced' to list additional commands.
linphonec>
…

0 New

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The build repositories for what we doing is in

https://bitbucket.org/account/user/combcommunications/projects/CV4

Clone

https://bitbucket.org/combcommunications/yocto

all the others are submodules or look at the commit logs the linphone software yocto layer is from linphone.

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For some or other reason it does not like building the kernel i had to use the code from the sierra tarball ill look into it and make it work.

Hi,

Those links above don’t seem to work.

Is the source code for the VoIP service available?