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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
I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . I'm unsure about the details, but the sparse documentation for PJSIP suggests it should be possible through the pjsua_set_null_snd_dev() call.
Calls — PJSIP Project 2.10 documentation
https://docs.pjsip.org/en/latest/pjsua2/call.html
Application can update the call setting (e.g: add audio/video), or enable/disable codecs, or update other media session settings from within the callback, however, as mandated by the standard (RFC3261 section 14.2), it must ensure that the update overlaps with the existing media session (in codecs, transports, or other parameters) that require ...
Asterisk 13 Configuration_res_pjsip - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip
If set to yes, chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio. If set to no, chan_pjsip will send a 180 Ringing when told to indicate ringing and will NOT send it as audio. call_group
⚡ Sip Audio Session - SIPThor
https://docs-new.sipthor.net/w/sip_clients/sip_audio_session/
(y/n) Audio session established using "speex" codec at 16000Hz Audio RTP endpoints 192.168.1.124:50378 <-> 85.17.186.6:58868 RTP audio stream is encrypted Remote SIP User Agent is "Blink-0.9.0" Remote party has put the audio session on hold Audio session is put on hold Audio session ended by remote party Session duration was 6 seconds 2009-08 ...
PJSIP - Open Source SIP, Media, and NAT Traversal Library
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, …
sipsimpleclient-example/sip-audio-session at master ...
https://github.com/amakukha/sipsimpleclient-example/blob/master/sip-audio-session
description = 'This script can sit idle waiting for an incoming audio session, or initiate an outgoing audio session to a SIP address. The program will close the session and quit when Ctrl+D is pressed.'. parser. add_option ( '-a', '--account', type='string', dest='account', help='The account name to use for any outgoing traffic.
Asterisk PJSIP Troubleshooting Guide - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide
The res_pjsip_endpoint_identifier_anonymous.so module is responsible for matching the incoming request to the anonymous endpoint. If SIP traffic that you expect to be matched to the anonymous endpoint is being rejected, try the following troubleshooting steps: Ensure that res_pjsip_endpoint_identifier_anonymous.so is loaded and
Problem: Couldn't negotiate stream 0:audio-0:audio ...
https://community.asterisk.org/t/problem-couldnt-negotiate-stream-0audio-sendrecv-nothing/88431
[Apr 29 22:41:11] ERROR[2650]: res_pjsip_session.c:936 handle_incoming_sdp: 6007: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) Can someone help me to resolve this. TLS-Problem: Couldn’t negotiate stream 0:audio-0:audio:sendrecv. jcolp April 30, 2021, 8:52am #2. Your SIP trace is incomplete and does not include any part of a ...
What this error mean which I get while AGI - Asterisk SIP ...
https://community.asterisk.org/t/what-this-error-mean-which-i-get-while-agi/87206
pjsip.conf pjsip.txt (928 Bytes) The agi file which I try to implement maybe it could help, (Note: I can’t post .agi file here so I change file extension to .txt so please take it as note) speech-recog.txt (8.6 KB) One last extensions.conf I thought It’ll link all thing because I spent most of my time here in this file extensions.txt (373 ...
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