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Patch to disable PortAudio ALSA support in ... - pjsip
https://trac.pjsip.org/repos/ticket/116
Summary changed from Patch to disable ALSA support in configure script to Patch to disable PortAudio ALSA support in configure script TeluuCon? 2018: PortAudio? support has been removed. Note: See TracTickets for help on using tickets.
C - how to resolve this error while using pjsip? - Stack ...
https://stackoverflow.com/questions/20312927/c-how-to-resolve-this-error-while-using-pjsip
Most Linux systems run PulseAudio over Alsa, like yours (you can see this in GStreamer's logs) but pjsip by default enables PortAudio driver for Linux. To fix it, you can enable an available Alsa driver by adding:
Media Ports Framework (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__PORT.htm
PJMEDIA_PORT_DISABLE TX or RX is disabled from the port. It means get_frame() or put_frame() WILL NOT be called for this port. PJMEDIA_PORT_MUTE TX or RX is muted, which means that get_frame() or put_frame() will still be called, but the audio frame is discarded. PJMEDIA_PORT_ENABLE Enable TX and RX to/from this port.
Custom Query – pjsip Open source SIP, media, and NAT ...
https://trac.pjsip.org/repos/query?status=closed&page=7&order=component&row=description
Patch to disable PortAudio ALSA support in configure script: bennylp enhancement normal Known-Issues-and-Ideas: pjmedia ... PJSIP fails to set the SDP Marker flag to TRUE on the very first packet of each sequence as requested by RFC 2833 section 3.4.
#1636 (Add BDsound IMproved Audio Device ... - trac.pjsip.org
https://trac.pjsip.org/repos/ticket/1636
Since the bdIMAD library provide itself the echo cancellation and the latency management, is necessary to disable these features in the PJSIP librariy applications. For example in PJSUA sample application there is the need to provide the following commands: --ec-tail=0 --no-vad --capture-lat=0 --playback-lat=0.
Asterisk 13 Configuration_res_pjsip - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip
active - res_pjsip will make a connection to the peer. passive - res_pjsip will accept connections from the peer. actpass - res_pjsip will offer and accept connections from the peer. dtls_fingerprint. This option only applies if media_encryption is set to dtls. SHA-256; SHA-1; srtp_tag_32. This option only applies if media_encryption is set to ...
PJSUA-API Media Manipulation (2.10) - PJSIP - Open Source ...
https://www.pjsip.org/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm
PJSUA has rather powerful media features, which are built around the PJMEDIA conference bridge. Basically, all media "ports" (such as calls, WAV players, WAV playlist, file recorders, sound device, tone generators, etc) are terminated in the conference bridge, and application can manipulate the interconnection between these terminations freely ...
Conference Bridge (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__CONF.htm
The level adjustment is calculated with this formula: output = input * (adj_level+128) / 128. Using this, zero indicates no adjustment, the value -128 will mute the signal, and the value of +128 will make the signal 100% louder, +256 will make it 200% louder, etc.
Manual of pjsua - Command Line SIP User Agent/Softphone
https://www.pjsip.org/pjsua.htm
pjsua is an open source command line SIP user agent (softphone) that is used as the reference implementation for PJSIP, PJNATH, and PJMEDIA. Despite its simple command line appearance, it does pack many features! Mutiple lines/identities (account registrations). Multiple calls. IPv6 (added in version 1.2)
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