Pjsip No Sound. 6 and pjsip built and running together nicely without any app
6 and pjsip built and running together nicely without any apparent errors. When running pjsua 21:40:30. WAV sound using the PJSIP lib At first, I've though that the problem was with the PBX (freeswitch), but the problem didn't occured on SIP phones like Grandstreams. Mar 14, 2023 · Describe the bug Im try to integrate build success but when i make call: Make outbound call can hear voice Receive inbound call, can not hear sound view log i see Oct 26, 2023 · [Nov 16 04:40:42] ERROR[1199682]: res_pjsip_session. My PBX is a bit special as it is sitting on the router and it sees the LAN and WAN Apr 17, 2017 · The incoming call works normally while no sound is heard when making an outgoing call. 16. The same thing for the sip trunk, when I call from the outside the phone also rings but no sound at all. 1 Lollipop, I do re No audio is heard by remote party Checklists: Check that correct device is used Check that microphone is functioning properly by looping-back microphone to speaker device. Steps to fix: Use the status code provided by pjsip (pjsip_status_. Checking the quality of the sound device Check for audio underflows/overflows Check that correct device is used Check by looping back microphone to speaker Check if RTP packets are received Testing and optimizing audio device with pjsystest Checking by playing a WAV file Check for problematic clock rate Check for network impairments of incoming Feb 6, 2023 · I finally moved my FreePBX to pjsip. Use STUN and/or TURN and/or ICE One of the obvious reason why no Feb 25, 2020 · PJSIP and RingCentral — Part 2: Handle Audio Medias Welcome to the part 2 of the PJSIP and RingCentral article series! If you haven’t done so, please read part 1 first. pjsip. In other words, they are mute – sort of speak. to connect the call's media to sound device. In terms of the sound going in and out, you have no end nodes in your description, so where are you expecting to originate the sound and where to listen to it? Mar 28, 2016 · Is there some property that I need to set up in pjsip (pjsua) or in AudioToolbox library to enable a sound be played during a sip call? I know this is possible (Bria has this, Groundwire also, not sure if they are using pjsip to implement sip). 15845 [2020 Mar 6, 2023 · I think the logs will also need pjsip set logger on setting. ilbc_mode iLBC mode (20 or 30). The res_pjsip_endpoint_identifier_anonymous. No audio is heard by remote party Checklists: Check that correct device is used Check that microphone is functioning properly by looping-back microphone to speaker device. A debugging info will Jun 27, 2021 · OS, Distribution & Version:iOS14 PJSIP version: 2. conf. It worked well and I used the python script to convert sip. Check by looping back microphone to speaker. MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. 5 mm audio socket. It doesn't matter how I try and make a sound, I just get si… I have a computer that I want to make out-going calls from. local loopback). Hi, Has *anyone* gotten pjsua2 in pjsip 2. Please follow general sound device troubleshooting for your operating system. There's also a new diagram explaining media flow. But the issue is that there is no audio after the call is answered. This is the extension I am testing with. The soft phone correctly registers. 4. Below are general guide to get around the NAT problem. Seriously, what is Understanding Audio Media Flow Table of Contents Understanding Audio Media Flow Introduction Audio playback flow (the main flow) Audio recording flow Sound device timing problem Incoming RTP/RTCP Packets Introduction During a call, media components are managed by PJSUA-LIB, when PJSUA-LIB or PJSUA2 is used, or by the application if the application uses low level PJSIP or PJMEDIA API directly Feb 4, 2019 · If you are using Anveo retail, you are already connecting to them on port 5010 and the port number at your end can be anything, because it’s using registration. Aug 2, 2014 · 3] Compile the whole PJSIP project 4] Copy bdIMADpj. Would anyone know the issue here? “Lack of audio RTP activity” Thanks! Oct 25, 2019 · Solved No Audio when connecting to FreePBX PJSIP Thread starter denniszitha Start date Jun 22, 2021 Status Not open for further replies. /pjsua --null-audio --id=sip:***@ip --registrar=sip:server --realm=* --username=user --password=pass sip:phone#@server makes the calls very nicely, but of course I get no sound. One way to inspect which sound device is used is by setting the log level to 5 (--app-log-level=5 argument with pjsua). Jun 18, 2020 · 3- I was previously using an older version of PJSIP with PortAudio which had the capability of detecting default device change during a call and switching device accordingly. It facilitates high quality VoIP calls (p2p or on regular telephones) based on the open SIP protocol. Checking by playing a WAV file Download MicroSIP, full or lite version, installer or zip archive with portable version. I'm unsure about the details, but the sparse documentation for PJSIP suggests it sho It's also possible to play . It does work when I use an undefined variable in startTransmit() Steps to reproduce I'm trying Here the list of my devices: Audio Device List Found 6 devices 0: PA [bcm2835 ALSA: bcm2835 ALSA (hw:0,0)] (0/2) 1: PA [bcm2835 ALSA: bcm2835 IEC958/HDMI (hw:0,1)] (0 Apr 12, 2022 · Describe the bug I am implementing voice call using pjsip pjsua2 android sample app. Turning sound device ON Sep 15, 2024 · 文章浏览阅读2. Feb 6, 2023 · I finally moved my FreePBX to pjsip. http://paste. I did noticed the following when comparing the logs between the working pjsua and non-working pjsua2_demo. If I enable direct media, I’m able to hear one-way. If using a device on the local network I have sound, but If I register an external device which I’ve successfully registered I do not (have sound). Server is not behind a nat, endpoints (clients) are. May 19, 2020 · Do you refer to this : fwconsole pjsip set logger on ? Because i use PJSIP and i can confirm you tomorrow (when at office) about Asterisk Settings (external address and local networks) you Ask. Aug 2, 2020 · Everything worked perfectly on chansip. If they call out side via trunk it works well. So it has nothing to do with the pjsip libraries, but something else. Nov 13, 2014 · sudo apt-get install libasound2-dev Pay attention that pjsip would still fail to set the default audio device since you have done the make as this package was missing. While all internally, local phones work, phones that are remote and outside of the local network aren’t transmitting audio to and from. g: patches from issue/PR xyz] configure script params: config_site. Check that no other application is using the sound devices. If either is a CHANSIP extension there is no problem. I am completely new to asterisk yet I have managed to set up the server with the service and it runs smoothly among LAN users and it Jul 15, 2016 · We're developing an application for some embedded hardware which doesn't have yet any audio devices. Understanding Audio Media Flow Table of Contents Understanding Audio Media Flow Introduction Audio playback flow (the main flow) Audio recording flow Sound device timing problem Incoming RTP/RTCP Packets Introduction During a call, media components are managed by PJSUA-LIB, when PJSUA-LIB or PJSUA2 is used, or by the application if the application uses low level PJSIP or PJMEDIA API directly Swinney C. 3k次。本文指导您如何在Ubuntu和CentOS系统中使用命令查询声卡信息，并提供重新编译PJ项目以找到语音设备的方法。适用于前端开发、后端开发、移动开发、游戏开发等领域的开发者。 sound-problem page of pjsip, I do hear the sound. Dec 25, 2021 · Howdy partners, I’ve mange to install and configure VitalPBX on a VM a using my ISP Router to forward the needed ports to use with the Public IP Address. The phone seems to be configured properly as it can initiate calls and rings when dialed, but there is no audio output when in call (neither the microphone is working). conf [global] type = global max_initial_qualify_time = 4 [transport-udp] type = transport protocol = udp bind = 0. 8, pjsua2 I use this code to get a call, in microsip can accept, but no sound the main code is #!/usr/bin/env python3 import sys import os import logging im PJSIP is a free and open source multimedia communication library written in C with high level API in C, C++, Java, C#, and Python languages. I'm trying to make an outgoing call with pjsua_call_make_call. Jul 4, 2017 · When a 180 (Ringing) is received without a 183 (Session Progress) the phone does not emit any ringback tone thud giving no feeedback about the status of the ongoing call. 6. Call’s AudioMedia, to transmit and receive audio to/from remote person. Follow the guide: Test the sound device using pjsystest. Introduction to clock concept Master Port Sound Device Port Sound Device (Deprecated) Video media port Clock Generator Codec Framework Codec Registration Codec constants Audio Codec Framework Video Codec If no, the RTP/RTCP sockets will share the same ioqueue as SIP sockets, and no worker thread is needed. The trunk is a bit special as it uses tcp for SIP, a proxy, and requires NAPTR and SRC to work for the name resolutions. Mar 30, 2020 · Dialing out works with no issues, however when getting calls on the household trunk (PJSIP) the caller can not hear me until they speak or make some noise. 0. The issue on exists on internal calls. We’ve set up the external address inside the config, as well as the internal network. I checked the logs and find the application does not set the sound device when making an outgoing call. I set up a phone in Germany, it is a Mitel 6863, It registered like no ones business. There is a dialing sound, but there is no audio trading at all. Mar 28, 2016 · Is there some property that I need to set up in pjsip (pjsua) or in AudioToolbox library to enable a sound be played during a sip call? I know this is possible (Bria has this, Groundwire also, not sure if they are using pjsip to implement sip). It does work very well without any issue on Android version 8,9,10. Here, We can not Jul 26, 2023 · Hi, I am having issue with the asterisk pjsip. 5mm audio out? We have no need to take a mic input and so just are looking to have PJSIP as a dumb client accepting an inbound call and outputting the incoming audio stream via the 3. c:937 handle_incoming_sdp: 2001: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) == Spawn extension (from-internal, 100, 4) exited non-zero on 'PJSIP/2001-00000000' May 5, 2016 · No Audio with SIPSTATION and SIP/PJSIP configured munozj (munozj) May 5, 2016, 6:38pm 1 Oct 31, 2018 · Does pjsua forcely check sound device? We just want to make call and transmit external DSP audio data to server, and we have already implemented fetching audio data from local DSP. It implements standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. Oct 25, 2010 · When on a call VoIP and receive a regular mobile call, if I pick up the regular call the application using VoIP goes to the background and when I switch back to the VoIP call, there is no audio. The above output shows no media flow between call and sound device. jb_max Jitter buffer maximum size in milliseconds. Oct 26, 2016 · When calling from an XLite softphone to a Callcentric number which has an Asterisk PJSIP channel registered, we cannot hear anything at all on the softphone (though the call is indeed established). Steps to fix: Use the status code provided by pjsip (pjsip_status_ The sound tests of PJSIP returns without any problems (we can even play sound twice). The sound device may be inactive if the application has set the auto close feature to non-zero (the snd_auto_close_time setting in pjsua_media_config), or if null sound device or no sound device has been configured via the pjsua_set_no_snd_dev () function. Sep 12, 2020 · Describe the bug Fail to detect audio devices. Jan 19, 2019 · I recently set up my asterisk 13 with PJSIP and database. 2. h contents: #define PJ_CONFIG_IPHONE 1 #define PJME Getting around NAT (for media) Table of Contents Getting around NAT (for media) Use STUN and/or TURN and/or ICE Disabling VAD Using Port Forwarding Still having NAT problem? Problems with NAT will cause no packets are getting received, either by local or remote party. Please let us know if these guides prove to be useful, or if they just miss the point completely! Jan 8, 2020 · No voice transmission, PJSIP behind NAT Asked 6 years ago Modified 4 years, 11 months ago Viewed 5k times Oct 4, 2022 · Describe the bug Playing a wav file in a call doesn't seem to work when the null audio device is set. When ICE is used, this callback will also be called to report ICE negotiation failure. pjsip edited Oct 25, 2010 at 15:11 Michael Eakins 4,18033754 asked Oct 25, 2010 at The JNI sound device device implementation was kindly donated by Regis Montoya from his CSIPSimple project. Upon creating PJSIP extension, I get no audio at all. Everything is fine, but I am not hearing ringing sound when I am calling some one. To Reproduce Launch pjsystest-armv6l-unknown-linux-gnueabihf Expected behavior List of detected audio devices Logs/Screenshots Fail Audio Device Detec Sep 21, 2016 · Hey there, We’ve come up to a problem where outgoing calls to PSTN doesn’t let through incoming sound. Dec 13, 2017 · After investigation, AudDevManager::setPlaybackDev/setCaptureDev() always sets mode to PJSUA_SND_DEV_NO_IMMEDIATE_OPEN, so PJSUA simply saves the new sound device IDs for future device open without reopening currently active sound device. Please advise. 010 pjsua_app. Now I am getting no audio with 14. However, Specific Guides Audio troubleshooting checklists Testing and optimizing audio device with pjsystest View page source Apr 17, 2017 · The incoming call works normally while no sound is heard when making an outgoing call. 5. Interesting about this problem is that exactly the same code works on Acoustic Echo Cancellation (AEC) Multichannel capable, supporting both built-in HW AEC and several software EC implementations such as WebRTC AEC3, Speex AEC, as well as our own echo suppressor. However, once they connect there is no audio. They can dial and connect. Sep 11, 2018 · I have been messing around with the latest version of distro 14, and have not had this issue with 13. 010 pjsua_aud. Dec 5, 2013 · I am getting this error while using pjsip. Therefore we have added a new guide for troubleshooting sound problems. That means: Call from PSTN to PBX, Pick up. I figure that this would be a NAT/Firewall issue. Any ideas what could cause this problem? Any help would be much appreciated. All working fine, but sometimes I get no voice, where most of the time I get a voice. I wasn't sure it was my laptop sound device problem, because I tried all the drivers, and the application did appear in the "window sound mixer". In other words, calls between 2 PJSIP extensions in location A will work fine but between A and B there is no audio but the call will complete. It is common to not be able to use sound device when other application is using the device. Dec 18, 2019 · I have no trunk set up so i don’t care if someone hacks into it. Playback device’s AudioMedia, to play audio to the sound device. g. It's working, but when I answer this call on a device, I can't hear any sound. but on Android version 5. Jan 27, 2023 · Asterisk Support 18 814 January 7, 2008 Audio problems from external locations Asterisk Support 7 278 February 5, 2009 2 NAT + asterisk = no sound Asterisk Support 0 260 August 26, 2008 Inbound SIP no audio [outbound and zap r great] Asterisk Support 2 275 March 7, 2006 Problems with audio Asterisk Support 5 498 May 2, 2007 Sep 20, 2023 · the main environment is docker, python3. ) as not revealed any problems. I have all of the necessary codecs installed and I tried everything Zoiper’s site suggested, so I believe I need to configure something on Asterisk’s end. Dec 27, 2021 · Easy Guide: How To Configure NAT For PJSIP Endpoints chan _pjsip is no more NAT aware than chan_sip in terms of nat=*. it is a virtual machine and has no microphone/speakers (as it's a command line server). But other end, he is receiving call. Chan_sip works perfectly just not chan_pjsip Feb 13, 2019 · Hi! I started to use pjsip to connect to a SIP trunk (German Telekom). Asterisk 16 LTS & PJSIP; hello world works but no sound coming from endpoints (2 Solutions!!) - YouTube Nov 12, 2021 · Describe the bug Hello everyone, we currently have the problem of hearing no sound during a SIP-Call (we cant play WAVE-file), using PJSUA2 for Java on a Raspberry Pi. No audio is heard in local speaker Checklists: Check that correct device is used Check that no other application is using the devices. ubuntu. Except, no sound and the clients don't hangup at the end. See: Acoustic Echo Cancellation API WebRTC AEC3 support: #2722 (iOS, Android, Mac/Linux/posix), #2775 (Windows) Main webrtc integration: #1888 Hardware AEC/VPIO: #1778 Speex AEC: #589 See also WebRTC MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. I'm using pjsua to connect to our asterisk server, so specifying: . e. But unlike SIP_Chan, I am unable to set NAT Dec 12, 2020 · When I call for example from extension 204 to 207, the phone rings but when I pick it up no sound. More media objects may be added in the future. Because PJMEDIA has no thread, a “clock” must be provided to make the media frames flow inside the media pipeline in a timely manner. Apr 26, 2016 · No sound coming out from the speaker or going through the mic. 42. AudioMediaRecorder, to record audio to a WAV file. The RTP Ports has been No audio is heard in local speaker Checklists: Check that correct device is used Check that no other application is using the devices. . Playing/recording sound using ALSA (aplay, arecord etc. But when I change codec to ulaw it works fine and also when I change chan_pjsip to chan_sip, direct media using opus works fine. com/6504337/ /* Create audio device paramet Jun 20, 2015 · I have implemented a project for VOIP using PJSIP(PJSUA2). So I need RTP software? following is detail log, I am Apr 10, 2022 · I used PJSIP PJSUA API to develop iOS VoIP applications. With 13, all I did was create the extension , plugged in a telephone and I was good. Set sound device: capture=-1, playback=-2 21:40:30. It covers common audio issues including dropouts, noise, jitter, and acoustic echo cancellation problems, along with diagnostic procedures and solutions. Introduction to clock concept Master Port Sound Device Port Sound Device (Deprecated) Video media port Clock Generator Codec Framework Codec Registration Codec constants Audio Codec Framework Video Codec MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. I moved to PJSIP and I can’t hear audio on any of my calls. Currently, the only workaround is to use PJSIP’s Android JNI sound device instead (one way to do this is by defining PJMEDIA_AUDIO_DEV_HAS_ANDROID_JNI to 1 and PJMEDIA_AUDIO_DEV_HAS_OPENSL to 0). Then when the the sound device is opened, for example when a call is established, it should display something like this: Sep 7, 2021 · Hello all, I currently have 2 VM’s set up for calling each other, and they do so just fine. 0 If no audio is heard with both pjsua and playfile Chances are other apps are unable to play to that sound device either. Common issues when developing on Windows Table of Contents Common issues when developing on Windows Troubleshooting crash problem on Win32 Troubleshooting crash problem on Win32 Building Application with Debugging Info The best way to find the crash is to equip your program with debugging info (for the Release mode) so that we can know exactly where the crash location is. So next I will copy the program output and the way I initialize pjsua and play the audiofile. If SIP traffic that you expect to be matched to the anonymous endpoint is being rejected, try the following troubleshooting steps: Feb 27, 2020 · Hi I’ve FreePBX 15 with Asterisk 16. How to resolve this? i have mic/speaker in the system but its failing to get the device. I receive the call, but I don’t get any voice. conf to pjsip. They are calling each other over PJSIP, and both are capable of doing the echo test. So you need to build Pjsip once again at pjsip directory do the following respectively : Nov 8, 2015 · Finally got Asterisk 13. 12 years ago Hi All, Is anyone aware of a way to get PJSIP to work on the Raspberry Pi so that it might just use the 3. Some of the problems may include: the speaker is not working properly the level is set too low the WAV file contains blank recording Check that correct device is used Some audio problems occur simply because the wrong device is being used by the application. However, when I initiate a call, I get the following error: 19 Jun 25, 2021 · I've got a problem with pjsip. Jan 21, 2020 · In the previous article, you learned how to configure the PJSIP channel driver to connect a simple softphone client with your Asterisk installation. Sep 15, 2017 · I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . The call is still in established state but the audio is not there. I am able to call it and the call plays out, but there May 22, 2025 · Audio Issues Relevant source files This page provides systematic troubleshooting approaches for audio-related problems in PJSIP applications. However, I can see an ic Jul 30, 2019 · I have recently set up an Asterisk server with version 16. In- and outgoing signalling is also no problem, but there is no audio with these calls. AudioMediaPlayer, to play WAV file (s). For performance optimization of audio systems, see Performance Tuning. Capture device’s AudioMedia, to capture audio from the sound device. But unlike SIP_Chan, I am unable to set NAT Overview Asterisk currently contains two SIP stacks: the original chan_sip SIP channel driver which is a complete standalone implementation, has been present in all previous releases of Asterisk and no longer receives core support, and the newer chan_pjsip SIP stack that is based on Teluu's "pjproject" SIP stack. When I hang up the other phone doesn’t seem to know this either, what’s going on?? And yes, I configured local network on Asterisk sip settings I’m using FreePBX 15. 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. I successfully compiled the PJSIP iOS library and registered it successfully. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. Any idea why? Update: I swapped ports for chan_sip and PJSIP and started getting audio for PJSIP. 5 to work with audio? I've still had no luck. max_media_ports Specify maximum number of media ports to be created in the conference bridge. I Jun 17, 2016 · I cannot playback any wav files from the dialplan or get any audio at all with asterisk 13 pjsip. Oct 22, 2024 · I have PJSIP working now, for the most part, as far as connecting and registering goes. PJSIP config: removed RTP debug shows no log, nothing. In this section, we will configure and build PJSIP as a native library for Android, and PJSUA2 API Java/JNI interface that can be used by Android Java and Kotlin applications. Just no audio. c . Normal application would need to implement this callback, e. At the moment, we're simply evaluating whether PJSIP runs okay on the hardware and can send audi PJSIP PJSUA python no sound but call OK on Raspberry Ask Question Asked 6 years, 11 months ago Modified 6 years, 11 months ago A useful test to check whether the local sound device (capture and playback device) is working properly is by transmitting the audio from the capture device directly to the playback device (i. dll into the folder containing the PJSUA executable (/pjsip-apps/bin) 5] To start the PJSUA application be sure to disable the following feature in PJSIP application, passing to PJSUA executable the following command line arguments: --ec-tail=0 --no-vad --capture-lat=0 --playback-lat=0 Dec 7, 2006 · One of the most frequent problem apparent on the mailing list are related to sound. Internal calls are no problem. 10 applied patch(es): [e. PJMEDIA support Oboe audio capture and playback device. 0 LTS. For detailed information May 23, 2024 · The problem is only present on PJSIP extensions connecting from some public IP addresses. bidirectional communication is working fine! Call from PBX to PSTN, Pick up, voice from PBX to PSTN is running, from PSTN to PBX not. Android Oboe From Oboe GitHub page: “Oboe is a C++ library which makes it easy to build high-performance audio apps on Android”. Asterisk is deployed in kubernetes, we opened the ports 5060 for outbound communication to SIP servers and 10000-20000 ports for inbound communication from SIP servers. It simply breaks the sub-options of nat= into fully-fledged options, so that nat=comedia And for PJSIP and SIP this Video gives you good hints: Keep us updated! Oct 17, 2024 · call between chan_pjsip endpoints using direct media and codec as opus, has no audio. so module is responsible for matching the incoming request to the anonymous endpoint. So my device is definitly working properly, the pjsip executable is working, so that are all not the problem. The command below will establish unidirectional media flow from the sound device to the call: Because PJMEDIA has no thread, a “clock” must be provided to make the media frames flow inside the media pipeline in a timely manner. Check that speaker is functioning properly by looping-back microphone to speaker device. Here is my PJSIP configuration: No sound on analog phone connected to a VoIP modem as PJSIP extension Trying to set an analog phone connected to an optical VoIP modem (Huawei HG8145V5) as a PJSIP extension. I’ve two extensions registered as PJSIP, when they call each other, there is no audio. jb_min Jitter buffer minimum size in milliseconds. The only remaining issue is that there is no sound. Identify the sound problem and troubleshoot it using the steps described in: Checking for sound problems. With AnveoDirect, you connect to them on port 5060, but the port number on your end can be different, you just need to specify it correctly on their portal “Configure Destination Trunks”.