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Audio Device API (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__audio__device__api.htm
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.
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
Audio Device API Reference (2.10) - pjsip.org
https://www.pjsip.org/pjmedia/docs/html/group__s2__audio__device__reference.htm
The audio device supports setting echo cancellation fail length. The value of this capability is an unsigned integer representing the echo tail in milliseconds. The audio device has voice activity detection feature. The value of this capability is a pj_bool_t …
PJSIP and RingCentral — Part 2: Handle Audio Medias | …
https://medium.com/ringcentral-developers/pjsip-and-ringcentral-part-2-handle-audio-medias-c26b8ea61319
No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/77854
No Audio PJSIP. FreePBX. Endpoints. carminuch (Carminuch) 2021-09-08 09:15:06 UTC #1. Hello all, I currently have 2 VM’s set up for calling each other, and they do so just fine. However, once they connect there is no audio. They are calling each other over PJSIP, and both are capable of doing the echo test.
No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/65068
Normally, Asterisk relays audio between the parties. If A calls B, then A sends audio to Asterisk and Asterisk sends it to B, and vice-versa. However, when possible, pjsip attempts to get the parties to communicate directly. This reduces the load on the server, might save bandwidth charges and also reduces latency.
[pjsip] Audio problem
https://pjsip.pjsip.narkive.com/PZQgcBbG/audio-problem
pjproject/pjsip-apps/bin I can play an audio file using the commands:./pjsua --clock-rate=48000 --play-file=[file.wav] Then I issue the command cc 1 0 And I hear the sound file playing. However I noticed this as output: pjsua_app.c ..Turning sound device ON This is something I don't notice in my own application, there the only output I see is
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.
Media — PJSIP Project 2.10 documentation
https://docs.pjsip.org/en/latest/pjsua2/media.html
Media — PJSIP Project 2.10 documentation Media ¶ Media objects are objects that are capable to either produce media or takes media. An important subclass of Media is AudioMedia which represents audio media. There are several type of audio media objects supported in PJSUA2: Capture device’s AudioMedia, to capture audio from the sound device.
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