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change sip port and transport · Issue #305 ...

https://github.com/BelledonneCommunications/linphone-android/issues/305

audio_jitt_comp=60 video_jitt_comp=60 nortp_timeout=30 disable_upnp=1 [sound] playback_dev_id= ringer_dev_id= capture_dev_id= dtmf_player_amp=0.1. #remove this property for any application that is not Linphone public version itself #ec_calibrator_cool_tones=1 [misc] max_calls=10 history_max_size=100

linphonec working settings: my_hlam - LiveJournal

https://my-hlam.livejournal.com/44667.html

[email protected]

Linphone not pulling in custom config - WTware …

https://forum.wtware.com/viewtopic.php?t=48736

Code: Select all [sip] media_encryption=srtp use_ipv6=0 [rtp] audio_rtp_port=5004 video_rtp_port=0 nortp_timeout=0 audio_jitt_comp=60 video_jitt_comp=60 audio_adaptive_jitt_comp_enabled=1 video_adaptive_jitt_comp_enabled=1 [video] enabled=0 show_local=0 [audio_codec_0] mime=opus rate=8000 channels=1 enabled=1 [audio_codec_1] …

mediastreamer2: Creating typical VoIP audio streams.

https://download-mirror.savannah.gnu.org/releases/linphone/mediastreamer/doc/group__audio__stream__api.html

Typedefs: typedef struct _AudioStream AudioStream: typedef VideoStream VideoPreview: Functions: MS2_PUBLIC int audio_stream_start_full (AudioStream *stream, RtpProfile *profile, const char *remip, int remport, int rem_rtcp_port, int payload, int jitt_comp, const char *infile, const char *outfile, MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec): MS2_PUBLIC …

Make SIP (VoIP) calls with LinPhone from an ... - Arch Linux

https://bbs.archlinux.org/viewtopic.php?id=130809

No. I want a script that will generate a list of my Linphone contacts which are listed in the file ~/.linphonerc (each contact is located under a line with [friend_<NUMBER>])

linphone-iphone/linphonerc at master ... - GitHub

https://github.com/BelledonneCommunications/linphone-iphone/blob/master/Resources/linphonerc

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sip - Calling from 3rd party device (with "Videophone ...

https://stackoverflow.com/questions/30375746/calling-from-3rd-party-device-with-videophone-capabilities-to-linphonedeskt

I'm trying to write a proof of concept to show that a 3rd party hardware device with "Videophone" capabilities can call a web application on the same LAN. At the recommendation of the manufacturer...

Re: [Linphone-users] asterisk and linphonec setup - non-GNU

https://lists.nongnu.org/archive/html/linphone-users/2007-01/msg00059.html

Re: [Linphone-users] asterisk and linphonec setup. Date: Mon, 22 Jan 2007 16:56:51 -0700. User-agent: KMail/1.9.5. I have also noticed a lack of audio levels in the configuration file for 1.5. 1.2 had *_lev= settings. Craig On Monday 22 January 2007 3:14 pm, Craig Matsuura wrote: > The call is being made to the asterisk server (via ext), in the ...

linphonec: making gsm the preferred codec?

https://linphone-users.nongnu.narkive.com/Q32CnANv/linphonec-making-gsm-the-preferred-codec

If you just set gsm as your first codec (in [audio_codec_0]) with enabled=1 will make linphone use it preferabily (if remote sip phone supports it). If the remote sip phone does not support it, then it can fallback to other codecs, in their order of preference. Simon

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