FreedomVoip: Difference between revisions
imported>Dallas.legan information on a proposed testing method |
imported>Bnewbold quick cleanup |
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Running VoIP ("voice over IP": internet telephony) over the free network makes sense. Free (as in freedom and beer) telephony for all. | |||
'''NOTE:''' these are very preliminary, rough sketch notes. Jump in! | |||
== Raw Notes == | |||
Using the [http://wiki.2600hz.com/display/docs/Whistle+Home Freeswitch Based Whistle platform] in our Network Operations Center and as part of our tower build. | |||
These are some notes for linphonec, the CLI client provided by the Debian linphone-nox package. The idea behind this information is to allow remote testing of an VOIP/SIP server with a headless computer, to determine the cause of some problems, to find if they are network/firewall related or more deeply in the build/configuration of the server program installation. (dallas.legan) | |||
When using: | |||
>soundcard use files | >soundcard use files | ||
Need to launch 'play ...file' soon after call begins, sseems to block, hunting for something to use for a microphone, and will not even record, therefore: | |||
>soundcard use files | >soundcard use files | ||
>record <infile> | >record <infile> | ||
| Line 31: | Line 20: | ||
>play <outfile> | >play <outfile> | ||
>............. | >............. | ||
Running the 'play' command ASAP after 'call'. See false bug report [http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2009-May/013712.html here]. | |||
'play files' should be 16bit pcm 8000Hz as per [http://www.mail-archive.com/linphone-users@nongnu.org/msg01220.html this link]. | |||
Suggest making such files via: | |||
ffmpeg -i sound.2.play.file -ar 8000 -ac 1 -f wav play.out.wav | ffmpeg -i sound.2.play.file -ar 8000 -ac 1 -f wav play.out.wav | ||
seemed to work; if no '-ac 1', sound was at about 1/2 speed. | |||
== Resources == | |||
* [http://chili.freenetworkfoundation.org/projects/on-net-voice-communication-engineering-and-delivery-group Chili project page] | |||
{{FNFProject}} | |||
Latest revision as of 04:39, 14 June 2012
Running VoIP ("voice over IP": internet telephony) over the free network makes sense. Free (as in freedom and beer) telephony for all.
NOTE: these are very preliminary, rough sketch notes. Jump in!
Raw Notes
Using the Freeswitch Based Whistle platform in our Network Operations Center and as part of our tower build.
These are some notes for linphonec, the CLI client provided by the Debian linphone-nox package. The idea behind this information is to allow remote testing of an VOIP/SIP server with a headless computer, to determine the cause of some problems, to find if they are network/firewall related or more deeply in the build/configuration of the server program installation. (dallas.legan)
When using:
>soundcard use files
Need to launch 'play ...file' soon after call begins, sseems to block, hunting for something to use for a microphone, and will not even record, therefore:
>soundcard use files >record <infile> >call sip:.......... >play <outfile> >.............
Running the 'play' command ASAP after 'call'. See false bug report here.
'play files' should be 16bit pcm 8000Hz as per this link.
Suggest making such files via:
ffmpeg -i sound.2.play.file -ar 8000 -ac 1 -f wav play.out.wav
seemed to work; if no '-ac 1', sound was at about 1/2 speed.
Resources
| Free Network Infrastructure Projects (edit) | |
|---|---|
| Box - Node - Tower - Tunnel - Link Network Operations Center - Lab - VoIP - Stack - Overview |