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Support for Voice Processing IO Audio Unit on Mac - PJSIP
https://trac.pjsip.org/repos/ticket/1778
Voice Processing IO Audio Unit is considered to provide nice features such as echo cancellation & AGC, it is supported since Mac OSX 10.7. Our internal testing indicates that the EC performance is far more superior than any software EC. However, VPIO seems to work only with specific devices, thus requiring you to use the default/built-in audio ...
PJSIP and RingCentral — Part 2: Handle Audio Medias | …
https://medium.com/ringcentral-developers/pjsip-and-ringcentral-part-2-handle-audio-medias-c26b8ea61319
In order to create an AudioMediaPlayer, you need to specify a wav audio file: player.createPlayer ("source.wav", PJMEDIA_FILE_NO_LOOP); Pay attention to the PJMEDIA_FILE_NO_LOOP option. Without...
sip - How to get the audio stream from PJSIP when there …
https://stackoverflow.com/questions/46243029/how-to-get-the-audio-stream-from-pjsip-when-there-is-no-audio-hardware-device
If there is an hardware audio device driver installed the exact same code works fine on both OSes. I have compiled the PJSIP libraries with PJMEDIA_AUDIO_DEV_HAS_NULL_AUDIO enabled. On Linux the presence of the module snd-dummy does not help. How do I get access to the audio data stream from a SIP call after …
Distro and PJSIP: No Audio - FreePBX Community Forums
https://community.freepbx.org/t/distro-and-pjsip-no-audio/52675
The PJSIP port is for signaling, the RTP ports (which you aren’t changing) are for audio. If you were unable to register then sure, the PJSIP port didn’t work. But you’ve both said you made calls and there was no audio. Therefore PJSIP signaling port is working. Considering PJSIP’s default is 5060, some providers block this port.
PJSIP - Open Source SIP, Media, and NAT Traversal Library
PJSIP is both compact and feature rich. It supports audio, video, presence, and instant messaging, and has extensive documentation. PJSIP is very portable. On mobile devices, it abstracts system dependent features and in many cases is able to utilize the native multimedia capabilities of the device.
Unable to open sound dev · Issue #2749 · pjsip/pjproject ...
https://github.com/pjsip/pjproject/issues/2749
11:16:42.569 coreaudio_dev.c ...Using VoiceProcessingIO audio unit 11:16:42.731 pjsua_aud.c ..Unable to open sound dev The text was updated successfully, but …
Sound device management on MAC using CoreAudio · …
https://github.com/pjsip/pjproject/issues/2464
There are 4 devices reported by PJSIP as below Device 0, Built-in Microphone (capture=2, playback=0) Device 1, Built-in Output (capture=0, playback=2) Device 2, H800 Logitech Headset (capture=1, playback=0) Device 3, H800 Logitech Headset (capture=0, playback=2).
How to increase Microphone volume level in Android …
https://stackoverflow.com/questions/43861391/how-to-increase-microphone-volume-level-in-android-pjsip
From the PJSIP docs, you can see there is a method called that could adjust the signal level received. You can use it like the following, where volume is between 0 and 2.0. pjsua_conf_adjust_rx_level (0, volume); I saw in a few places that you might need root access to modify this parameter anyway, or that you need to have a MediaTek chip.
Media Quality — PJSIP Project 2.10 documentation
https://docs.pjsip.org/en/latest/pjsua2/media_quality.html
Identify the sound problem and troubleshoot it using the steps described in: Checking for sound problems. It is probably easier to do the testing using lower level API such as PJSUA since we already have a built-in pjsua sample app located in pjsip-apps/bin to do the testing.
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