If you are migrating from chan_sip to chan_pjsip, then also read the NAT section in Migrating from chan_sip to res_pjsip for helpful tips. On my endpoint, I am unable to hear anything but audio goes out. 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. Contribute to pjsip/pjproject development by creating an account on GitHub. The principle is very simple; application connects audio source to audio destination, and the bridge makes the audio flows from that source to the specified destination, and that’s it. Here is a sample code to post a job via schedule timer, in this sample, it is for scheduling a video capture device preview start. Looping Audio ¶ If you want, you can loop the audio of an audio media object to itself (i. 媒体（Media）媒体对象是能够产生媒体或接受媒体的对象。 Media的重要子类是AudioMedia，它代表音频媒体。PJSUA2支持多种类型的音频媒体对象：捕获设备的AudioMedia，用于从声音设备捕获音频。播放设备的AudioMedia，可以播放音频到声音设备。呼叫音频媒体， The above steps are okay for our simple purpose of changing file’s sampling rate. pjsip. The documentation is intended for developers building voice and video applications with PJSUA2. 14 version? Regards Establish unidirectional media flow to sink. Asterisk and Phones Connecting Through NAT to an ITSP So it is also recommended to avoid calling PJSIP API from GUI thread since: it may take some time to complete, or it may block while trying to acquire a lock. Is it building problem? Should I change to 2. But you can Apr 14, 2021 · I'm trying to play 16 bit PCM mono . media terminations may also have different frame time; the conference bridge will perform the necessary bufferring PJMEDIA Core Core PJMEDIA was designed to be applicable in broad range of systems, from desktop to mobile, embedded, and maybe even DSP. org [mailto:pjsip-***@lists. This source is an example to demonstrate using SIP and RTP/RTCP framework to measure the network quality/impairment from the SIP call. Step 1: Acquire an IP Phone First, you will need one or more VoIP phones. That means you can loop the call’s audio media, so that audio received from the remote person will be transmitted back to her/him. Specify maximum number of media ports to be created in the conference bridge. You can loop-back audio from any objects, as long as the object has bidirectional media. Nov 2, 2023 · I have implemented a subclass of the AudioMediaPort class and provided an override of the onFrameReceived () method. The behavior of this function depends on whether device or software AEC is being used. 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, embedded Jun 30, 2024 · static pjmedia_snd_port *g_snd_port; /* Sound device. void createPort(const string &name, MediaFormatAudio &fmt) Create an audio media port and register it to the conference bridge. And if multiple sources are transmitting to the same sink, the media will be mixed together. Interesting is also that the example application which gets created while building pjsip, runs without any problems on our Raspberry Pi (we can hear audio during call), so our device is potentially Jun 7, 2021 · I am currently on a VOIP project on Android, I am using PJSIP for my project, everything works quite well. Returns: PJ_SUCCESS on success. PJSIP WSS Transport Although the HTTP server does the heavy lifting for WebSockets, we still need to define a basic PJSIP Transport: Change the echo cancellation settings. AudDevManager &audDevManager() ¶ Get the instance of Audio Device Manager. Application can create a derived class and use registerMediaPort2 () / unregisterMediaPort () to register/unregister a media port to/from the conference bridge. conf is a flat text file composed of sections like most configuration files used with Asterisk. WAV", // file name Enums enum pjmedia_aud_dev_cap This enumeration identifies various audio device capabilities. The library will not The list of audio media port. Since all media terminate in the bridge (calls, file player, file recorder, etc), the value must be large enough to support all of them. 10). FEATURES - Session Initiation Protocol (SIP) features: - Basic registration and call - Multiple accounts - Call hold, attended and unattended call transfer - Presence - Instant messaging - Multiple SIP accounts - Media features: - Audio - Conferencing - Narrowband and wideband Sep 30, 2019 · 文章浏览阅读2. This file is pjsip-apps/src 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. media terminations may also have different frame time; the conference bridge will perform the necessary bufferring TransportId transportCreate (pjsip_transport_type_e type, const TransportConfig &cfg) PJSUA2_THROW (Error) Create and start a new SIP transport according to the specified settings. This is a lite wrapper class for audio conference bridge port, i. 7k次，点赞4次，收藏8次。本文深入解析音视频流的构成与处理流程，详细介绍了媒体流结构体pjmedia_stream的内部机制，包括流的创建、启动过程，以及关键回调函数的作用。通过示例程序simpleua. e PJ_IOQUEUE_MAX_HANDLES which by default is set to 64, and all of them are being MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. A master port has two media ports connected to it, and by convention thay are called downstream and upstream ports. pj_ssize_t Overview 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. VideoMediaVector mediaEnumVidPorts () const PJSUA2_THROW (Error) Enumerate all video media port. conf or pjsip. Note that the remote peer must support RTCP. status = pjmedia_wav_player_port_create(pool, "Input. Oct 24, 2025 · I have started a simple project that is capable of making a SIP establishment for voice call and then forwarding RTP packet through another endpoint. the audio received from the object will be transmitted to itself). It is called whenever a frame is received into the conference it is connected to. 9 in "nicolaipre/python3-pjsip-memory-buffer". pj_ssize_t Feb 18, 2025 · 文章浏览阅读1. VidDevManager &vidDevManager() ¶ Get the instance of Video Device Info: You can set the device settings when opening audio stream by setting the flags and the appropriate setting in pjmedia_aud_param when calling pjmedia_aud_stream_create () Info: Once the audio stream is running, you can retrieve or change the stream setting by specifying the capability in pjmedia_aud_stream_get_cap () and pjmedia_aud_stream_set_cap () respectively. I am using 2. 9k次。本文详细解析了PJSIP项目中音频混音的工作原理和技术细节，包括媒体流传递过程、音频混音分析等内容。 Contribute to VoIPGRID/PJSIP development by creating an account on GitHub. media termination can be connected to itself to create loopback media. Configure PJSIP If you're not already familiar with configuring Asterisk's chan_pjsip driver, visit the res_pjsip configuration page. These are the core considerations for such design: any clockrates N-channels support zero thread capable Audio Features Some audio processing algorithms implemented in PJMEDIA. Sep 14, 2023 · 文章浏览阅读2. Creating audio streams 16 years ago Hello, Could you post what did you do step by step ? Samuel Post by Maya Zalcberg Hi Samuel, "Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) ". I have put all the information under pjsip_custom. PJSIP project. the media termination may have different clock rates, and resampling will be done automatically by conference bridge. 我们在使用过程中，可能涉及到输入输出通道的切换或者重置，那么 pjsip 就原生提供了相应的接口，下面我们来分析下它的具体实现流程。前言开始分析输入输出的时候先普及下基础知识，如果已经了解过的可以不用看此处 Typedefs typedef struct pjsua_ext_snd_dev pjsua_ext_snd_dev Opaque type of extra sound device, an additional sound device beside the primary sound device (the one instantiated via pjsua_set_snd_dev () or pjsua_set_snd_dev2 ()). chan_sip is the legacy Asterisk SIP implementation. org PJSIP project. API The above steps are okay for our simple purpose of changing file’s sampling rate. This media port will act as a source, and it may transmit to multiple destinations/sink. But for other purposes, the process of reading and writing frames need to be done in timely manner (for example, sending RTP packets to remote stream). If SIP traffic that you expect to be matched to the anonymous endpoint is being rejected, try the following troubleshooting steps: Apr 17, 2017 · 6. g: connect it to/from other ports, adjust/check audio Sep 24, 2024 · Find answers to how to record the customer voice on pjsip from the expert community at Experts Exchange Nov 14, 2024 · Describe the feature How to use AudioMediaPort in andriod? How to use onFrameReceived and onFrameRequested? Can you give a sample for the developer? Thank you very much Describe alternatives you've Looping Audio ¶ If you want, you can loop the audio of an audio media object to itself (i. Translating ports in the router can cause various issues. Fork 0 master pjproject / pjsip-apps / src / samples /pjsua2_demo. e: this class only maintains one data member, conference slot ID, and the methods are simply proxies for conference bridge operations. The logs don't indicate any errors, however I don't hear anything on the other side. pj::Account (C++ class) pj::Account::Account (C++ function) pj::Account::create (C++ function) pj::Account::enumBuddies (C++ function) pj::Account::enumBuddies2 (C++ Parameters: port – The file player port. org - pjsip/pjproject_docs Dec 9, 2019 · I ported the customization to python3 and pjproject 2. The media transport (pjmedia_transport) is the object to send and receive media packets over the network. A media port interface basically has the following properties: media port information (pjmedia_port_info) to describe the media port Aug 7, 2023 · Hi all, I am creating a small (but overkill, let’s be honest) phone system for our home as our ISP has just switched us over to a sip-based landline rather than POTS. wav files in a call with PJSUA 2. Other features of PJSUA media: efficient N to M interconnections between media terminations. org] On Behalf Of Nanang Izzuddin Sent: Wednesday, 13 August 2008 8:16 PM To: pjsip list Subject: Re: [pjsip] Creating custom conf port in pjsua Hi Pierre, The limitation seems to come from ioqueue, i. Dec 20, 2025 · However, res_pjsip is considerably more complicated to set up than the older chan_sip. inline virtual void onFrameRequested(MediaFrame &frame) Jun 18, 2021 · This tutorial will walk you through configuring Asterisk to service WebRTC clients. Returns The list of video media port. The configuration below shows how each of these files are used when an outgoing autopatch or reverse autopatch call is made. These audio capabilities indicates what features are supported by the underlying audio device implementation. Group PJMEDIA_MASTER_PORT group PJMEDIA_MASTER_PORT Thread based media clock provider. k. 14. pj_status_t pjmedia_snd_port_create(pj_pool_t *pool, int rec_id, int play_id, unsigned clock_rate, unsigned channel_count, unsigned samples_per_frame, unsigned bits_per_sample, unsigned options, pjmedia_snd_port **p_port) Create Change the echo cancellation settings. 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. e. offset – Playback position in bytes, relative to the start of the payload. Detailed Description Media Port Concepts Media Port A media port (represented with pjmedia_port "class") provides a generic and extensible framework for implementing media elements. But you can Other features of PJSUA media: efficient N to M interconnections between media terminations. As explained in Portable Sound Hardware Abstraction, the sound hardware abstraction provides some callbacks for its user: it calls callback when it has finished capturing one media frame, and it calls when it needs media frame to be played to the sound playback hardware. But for a network requirement, I need to be able to configure the RTP ports for both audio and video, as can be done in other voip software like Linphone. py Note that currently the issue page is for bug report or feature request only. Class Reference ¶ Media Framework ¶ Classes ¶ Warning doxygenclass: Cannot find class “pj::Media” in doxygen xml output for project “pjsip” from directory: pjproject/pjsip/docs/xml Nov 12, 2021 · When I try to access the remote audio media (port to the person we called) for example by trying to get the port Id, the whole programm crashes. If the This example demonstrate how to create a custom media port (in this case, a sine wave generator) and connect it to the sound device. But you can Media/Audio Features Table of Contents Media/Audio Features Core Audio Features Video Features Transports Media components (Ports) Clock provider Codec Framework SDP RTP and RTCP Compile Time Settings Basic Types and Functions Endpoint Formats Media Flow Events Core PJMEDIA was designed to be applicable in broad range of systems, from desktop to mobile, embedded, and maybe even DSP. PJSUA2 wraps together the signaling, media, and NAT traversal functionality into easy to use call control API, account management, buddy list management, presence, and Other features of PJSUA media: efficient N to M interconnections between media terminations. a Voice over IP/VoIP softphones). At first, a registered audio media will not be connected to anything, so media will not flow from/to any objects. It changes the received Contact header to be the actual source IP address and port of the SIP request and effectively ignores what the other party stated. g: connect it to/from other ports, adjust/check audio Functions void pjmedia_snd_port_param_default(pjmedia_snd_port_param *prm) Initialize pjmedia_snd_port_param with default values. It contains 7 sections that describe key aspects of developing applications with PJSUA2, including development guidelines, the main PJSUA2 classes, managing endpoints, accounts, media, and calls. May 25, 2023 · SIP / PJSIP On some guides online, you will see references to the chan_sip and chan_pjsip modules. // Timer type ID enum { TIMER_START_PREVIEW = 1, The res_pjsip_endpoint_identifier_anonymous. And more over, as the application’s scope goes bigger, the same pattern of manually reading/writing frames comes up more and more often, thus perhaps it would pjmedia_port *player, *resample, *writer; pj_status_t status; // Create the file player port. Applications get these capabilities in the pjmedia_aud_dev_info structure. If application wants to be notified on playback EOF event, it can subclass pj::AudioMediaPlayer and implement pj::AudioMediaPlayer::onEof2() callback. The prevailing wisdom of the wider Asterisk community suggests to only use chan_sip if you already have an existing configuration for it, while also planning the upgrade to res_pjsip. Application can also set the specific features/capabilities when opening the audio stream by setting the flags member of pjmedia_aud_param structure Jan 1, 2020 · The chan_pjsip module provides the “rewrite_contact” option to overcome this. The media transport interface allows the library to be extended to support different types of transports to send and receive packets. c，展示了如何在SIP协商成功后创建并启动音频流，揭示了数据从网络到设备的完整路径。 Four configuration files will be used to setup the autopatch: rpt. org) 1. The media stream flowing to the downstream port is called encoding or send direction, and media stream flowing to the upstream port is called decoding or receive direction (imagine the Parameters: port – The file player port. It covers common audio issues including dropouts, noise, jitter, and acoustic echo cancellati Audio Media. The Sound Device Port Supported Audio Devices PJMEDIA-Audiodev supports the following platforms/devices: ALSA Android OpenSL (deprecated) Android JNI Android Oboe bdIMAD by BdSound CoreAudio (Mac OS X and iPhone) PortAudio WMME (Windows and Windows Mobile devices) WASAPI (Windows Audio Session API) Older devices that are no longer supported: Blackberry BB10 Symbian audio streaming/multimedia framework (MMF) Nokia This document provides documentation for PJSUA2, a high-level API for the PJSIP multimedia framework. conf, extensions. Creating audio streams Typedefs typedef struct pjsua_ext_snd_dev pjsua_ext_snd_dev Opaque type of extra sound device, an additional sound device beside the primary sound device (the one instantiated via pjsua_set_snd_dev () or pjsua_set_snd_dev2 ()). The conference bridge SIP Service for Android based on PJSIP. You should be using PJSIP for everything these days. conf, modules. This happens when media is removed or added during a call. Source and sink may refer to the same Media, effectively looping the media. Thanks for reply，so far my app always crushed when running to the new AudioMediaPort () step. I have disabl Download MicroSIP, full or lite version, installer or zip archive with portable version. argentek (Andy M) February 4, 2019, 4 Detailed Description PJSUA has rather powerful media features, which are built around the PJMEDIA conference bridge. Using a random port gives me one way audio. This will unregister the audio media port from the conference bridge. Nov 14, 2024 · Check the sample for C++ and Python, and try to do the same thing on Java. 媒体（Media）媒体对象是能够产生媒体或接受媒体的对象。 Media的重要子类是AudioMedia，它代表音频媒体。PJSUA2支持多种类型的音频媒体对象：捕获设备的AudioMedia，用于从声音设备捕获音频。播放设备的AudioMedia，可以播放音频到声音设备。呼叫音频媒体， May 9, 2018 · pjsip 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. The echo cancellation settings should have been specified when this sound port was created, by setting the appropriate fields in the pjmedia_aud_param, because not all sound device implementation supports changing the EC setting once the device has been opened. Returns The Audio Device Manager. It facilitates high quality VoIP calls (p2p or on regular telephones) based on the open SIP protocol. The layout of the program has been The conference bridge The conference bridge provides a simple but yet powerful concept to manage audio flow between the audio medias. I used PJSUA2 python library and right now I can From: pjsip-***@lists. I know it may not be of use to you for pjsua2, but maybe it can help someone else looking for something similar. Parameters: prm – The parameter. media terminations may also have different frame time; the conference bridge will perform the necessary bufferring An: ***@lists. Feb 3, 2025 · Check AudioMediaPort class and the sample at swig/python/test. Application can also set the specific features/capabilities when opening the audio stream by setting the flags member Without looping, silence will be played once the playback has reached the end of the WAV file. cpp sauwming c5bc3d1ef5 Add function to initialize MediaFormat audio & video (#3925) last year Specific Guides Audio troubleshooting checklists Check audio interconnection in the conference bridge View page source pjsip. The VOIP provider should provide a stanza for you to insert in iax. 2k次。本文深入探讨PJSUA-LIB库中发起呼叫的内部流程，详细解析pjsua_call_make_call函数的工作机制，包括呼叫标识分配、呼叫参数设置、SIP对话创建等关键步骤。 This enumeration identifies various audio device capabilities. Once application is done with the playback, just call pj::AudioMedia::stopTransmit() to stop the playback: Destructor. Media element itself could be a media source, sink, or processing element. conf. However, the larger the value, the more computations are performed. org pjsip mailing list http://lists. Regards, Maya _______________________________________________ Visit our blog: http://blog. */ static void call_on_media_update( pjsip_inv_session *inv, pj_status_t status) {} pjmedia_port *media_port; /* Get the media port interface of the audio stream. Nov 19, 2025 · Stack Overflow | The World’s Largest Online Community for Developers May 22, 2025 · This page provides systematic troubleshooting approaches for audio-related problems in PJSIP applications. This program can be used to make calls or to receive calls from other SIP endpoint (or other siprtp program), and to display the media quality statistics at the end of the call. For further questions, please use any other means such as StackOverflow. Unfortunately, they require some very specific settings and custom configurations which means it must be defined in a file rather than through the trunks tab on freePBX. . The media transport is declared as pjmedia_transport “class”, which declares “interfaces Configuring res_pjsip to work through NAT Overview Here we can show some examples of working configuration for Asterisk's SIP channel driver when Asterisk is behind NAT (Network Address Translation). Parameters: name – The port name. These are Download MicroSIP, full or lite version, installer or zip archive with portable version. Interesting is also that the example application which gets created while building pjsip, runs without any problems on our Raspberry Pi (we can hear audio during call), so our device is potentially Info: You can set the device settings when opening audio stream by setting the flags and the appropriate setting in pjmedia_aud_param when calling pjmedia_aud_stream_create () Info: Once the audio stream is running, you can retrieve or change the stream setting by specifying the capability in pjmedia_aud_stream_get_cap () and pjmedia_aud_stream_set_cap () respectively. Aug 14, 2023 · When you type ‘pjsip set logger on’ at the Asterisk command prompt, ‘PJSIP Logging enabled’ should be the response. Contribute to VoiSmart/pjsip-android development by creating an account on GitHub. Apr 17, 2017 · 6. org Betreff: [pjsip] Send audio wav only on left device channel Hi guys, I'm working on pjmedia split audio channel left/right and I have the following issue. 11 (also happened with 2. Group PJMED_SND_PORT ¶ group PJMED_SND_PORT Media Port Connection Abstraction to the Sound Device. conf Source and configuration files for https://docs. 1, is a issue or not. fmt – The audio format. And more over, as the application’s scope goes bigger, the same pattern of manually reading/writing frames comes up more and more often, thus perhaps it would This example demonstrate how to create a custom media port (in this case, a sine wave generator) and connect it to the sound device. so module is responsible for matching the incoming request to the anonymous endpoint. This file is pjsip-apps/src Specify maximum number of media ports to be created in the conference bridge. Acoustic Echo Cancellation Adaptive Delay Buffer Adaptive Jitter Buffer Adaptive PJMEDIA - Media Stack ¶ PJMEDIA is a fully featured open source media stack, featuring small footprint and good extensibility and excellent portability. 9k次，点赞9次，收藏13次。会议桥自定义媒体端口在《pjSIP注册呼叫流程简介》中介绍了pjSIP注册与呼叫的基本流程，本节对自定义媒体流与端口做下介绍。会议桥pjSIP中通过会议桥（Conference）把媒体流（Stream）与抽象音频设备端口（Sound Device Port）连接起来（并负责各路媒体的混流 Feb 4, 2019 · On the pjsip end, change the port to something else, forward that port and adjust the settings in your devices to connect to the new port. Looking at a larger section of the log, do you see the trunk becoming reachable and unreachable repeatedly? Or did it just go unreachable and stay that way? Table of Contents The Endpoint Accounts Working with audio media Working with video media Calls Presence and Instant Messaging library based on PJSIP stack (http://www. For any new installations, res_pjsip is recommended. If more than one sources are transmitting to the same destination, then the Note Be prepared that the transport adapter may be destroyed while the call is running, and/or the pjsua_callback::on_create_media_transport callback is called again for the same call (thus this callback may be called more than once for a call). org/mailman/listinfo/pjsip_lists. Apr 25, 2025 · Custom Audio Processing with AudioMediaPort For applications requiring custom audio processing, AudioMediaPort provides a way to intercept and modify audio frames: In PJSUA2, all audio media objects are registered to the central conference bridge for easier manipulation. Default value: PJSUA_MAX_CONF_PORTS Detailed Description Media Port Concepts Media Port A media port (represented with pjmedia_port "class") provides a generic and extensible framework for implementing media elements. If the Looping Audio ¶ If you want, you can loop the audio of an audio media object to itself (i. My source code is: //Test split left right channel pjmedia_snd_port Nov 12, 2021 · When I try to access the remote audio media (port to the person we called) for example by trying to get the port Id, the whole programm crashes. A media port interface basically has the following properties: media port information (pjmedia_port_info) to describe the media port 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. conf, and either iax. Each section defines configuration for a configuration object within res_pjsip or an associated module. Introduction to PJSUA2 PJSUA2 API is a C++ library on top of PJSUA-LIB API to provide high level API for constructing Session Initiation Protocol (SIP) multimedia user agent applications (a. I don't know how to send audio from wav audio file only on left channel of default my audio device. Default value: PJSUA_MAX_CONF_PORTS Group PJMEDIA_TRANSPORT group PJMEDIA_TRANSPORT Transports. Mar 14, 2025 · 文章浏览阅读2. This sound device is also registered to conference bridge so it can be used as a normal conference bridge port, e.

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Audiomediaport Pjsip. If you are migrating from chan_sip to chan_pjsip, then also read