Pjsip conference bridge. When the audio media becomes inactive (for example when the call is put on hold), there is no need to stop the call’s audio media transmission since they will be removed automatically from the conference bridge, and this will automatically remove all connections to/from the call. Contribute to pjsip/pjproject development by creating an account on GitHub. However you will then be limited to using a single processor core which might be a problem if you want to handle hundreds of simultaneous calls. This implementation was done using Ubuntu 20. A simple call will be total 240ms latency caused by Mar 3, 2016 · Can pjsua handle such load? > How much simultaneous calls can pjsip with pjmedia using a bridge for > each call handle? > Thx > On 3/2/2016 5:10 PM, Bill Gardner wrote: >> Hi Alaa, >> >> It's pretty easy to code an announcement system using pjsua, and if >> you don't want to use pjsua you can look at the pjsua code as a >> guide. Check audio interconnection in the conference bridge Use pjsua’s cl (conference list) command from the pjsua ’s menu to check if the connection is made between the call and the sound device in the conference bridge. What I have done is: Call A -> B Put B on hold Call A -> C Re-invite B The issue I am facing is, I can hear voice f conference calls on pjsua. 04 LTS-based Asterisk in a Triangle topology (browser-server-browser). Mar 30, 2015 · Based on my experience of using PJSIP on desktop, you should call all the parties with different calls to pjsua_call_make_call (execute pjsua_call_make_call 4 times for 4 accounts in group for example). I wonder why,does anyone has any idea about this and how could we increase the number of users conferencing Also if conference can be achieved by the following method mentioned above. k. 6. When the signal level changes significantly in the subsequent frame, this causes the subsequent frame to be misaligned with previous frame, and this would produce a noise. Mathieu Nanang Izzuddin a ?crit : > Glad to hear your feedback! > > In this case, when sound device/resampler asks for audio frame to the > conference bridge, non-audio frame type is returned by conference > bridge since no port is connected to sound device/resample port yet > (no transmitter). I check code but i suppose this is not all. Group PJMEDIA_CONF ¶ group PJMEDIA_CONF Audio conference bridge implementation. conference bridge and use this for all calls. res_pjsip Configuration Examples Below are some sample configurations to demonstrate various scenarios with complete pjsip. This describes the video conference bridge implementation in PJMEDIA. There are several types of video media objects supported in PJSUA2: library based on PJSIP stack (http://www. c模块是用来做音频设备和媒体数据流 Nov 12, 2021 · pjsua_aud. c pjmedia_conf_remove_port则从会议桥中移除一个port PJSIP version 2. pjsip. There are several types of video media objects supported in PJSUA2: PJSUA has rather powerful media features, which are built around the PJMEDIA conference bridge. 9k次。本文详细解析了PJSIP项目中音频混音的实现原理。重点介绍了conference模块如何管理多个音频流，并进行混音操作。同时，还探讨了master_port的作用及其实现细节。 As our conference bridge shouldn't be doing resampling or transcoding of any kind assuming we have the same audio codec of VoIP Provider and on Modem Server, we may just enable the Audio Switchboard of PJSIP designed to replace Conference Bridge by just doing RTP packet forwarding between the legs without Audio Latency. I am trying to use he suggestion at http://trac. Hi, I am trying to use he suggestion at http://trac. cpp pjsua_conf_connect (id, sink. Sometime try to connect player to media fails with following error: media. However, in Asterisk 15 the single pipe has been swapped for multiple ones. conf dial plan with minimal fuss. conference bridge. The conference bridge calls pjmedia_port_get_frame() for all ports in the conference bridge, then it mixes the signal together according to ports connection in the bridge, and deliver the mixed signal by calling pjmedia_port_put_frame() for all ports in the bridge according to their connection. g: via event PJMEDIA_EVENT_FMT_CHANGED), it needs to be updated by refreshing the port connections. It only happens when I connect the two streams (just adding them to the conf bridge isn't enough) and I've used other Sep 20, 2017 · For instance, only one flow of video is allowed, thus limiting its usefulness in conferencing applications. After calls are estabilished, you should connect them in PJSIP's conference bridge all-with-all with pjsua_conf_connect function. Jul 3, 2019 · 本文深入分析了Pjsip的Conference模块，探讨了如何抽象Port并实现混音功能。Port的核心操作包括put_frame、get_frame和on_destroy。conference通过管理port并利用delay_buf解决录音与播放同步问题。混音通过pjmedia_conf_connect_port接口实现，源Port的PCM数据在get_frame中与目标Port的mix_buf混合，完成多方通话的混音效果。 Feb 25, 2020 · The Audio Conference Bridge The conference bridge provides a simple but yet powerful concept to manage audio flow between the audio medias. PJSUA2 wraps together the signaling, media, and NAT traversal functionality into easy to use call control API, account management, buddy list management, presence, and There is another problem, specific to when upsampling is done in conference bridge. Can conference be implemented without conference bridge in PJSUA as media mixing can be done by my audio device? Also is the TICKET 1185 (ifdef DISABLED_1185 in pjsip code) would be helpful in this regard? 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. Apr 25, 2025 · PJSIP is a comprehensive, high-performance, and open-source multimedia communication library written in C. The topics are also shown on the navigation menu on Feb 14, 2020 · Main focus of this release is: Video conference Darwin (Mac & iOS) native SSL backend NAT enhancement: TURN over TLS SIP multiple TCP/TLS listeners Among several bug fixes and enhancements, it includes important updates such as improved thread safety in PJSUA2 list objects (ticket #2189) and updated build configs for newer Android NDKs. To see examples side by side with old chan_sip config head to Migrating from chan_sip to res_pjsip. It is also okey to make more than one call for different destinations. In my first approach I used the conf bridge to connect each stream to a custom pjmedia_port, but after the 5th-6th connection the cpu performance becomes poor and the timing gap between each put_frame (same Apr 11, 2023 · pjsip音频流传递过程和混音算法_lincai2018的博客-爱代码爱编程_pjsip混音 2018-03-23 分类: 音视频 pjsip 对于实现voip，pjsip是一个非常优秀的开源项目。其实现了复杂的sip信令交互和音频的传输建立。 1、媒体流的传递过程 我们来结合代码分析下媒体流的传递。 conference. Okey making multiple calls for same destination is cool. h) . 2008-08-07 14:41:53: @nanangizz created the issue on trac ticket 587 Currently the conference bridge will introduce delay to the audio flow. The * conference bridge provides powerful and very efficient mechanism to The conference bridge provides powerful switching and mixing functionality for application. Group PJMEDIA_CONF group PJMEDIA_CONF Audio conference bridge implementation. Those not connected via chan_pjsip will get whatever the channel driver supports. It would be more proper to compare baresip to PJSIP/pjsua. The reason I want to do this is that I am Aug 31, 2019 · When you make work to get an incoming call with the conference bridge, then try to use 2 threads to manage 2 different endpoints. I'm using both and both are great, with multiple unique features. Video conference bridge specifications Video timing clock, just like audio, the video conference bridge will have a clock that schedules video flow between ports. After hangup both conference ports are freed (I see this in mediaEnumPorts () output), when new call with new media and new player is made, it reuses previously freed ports on the bridge. PJSUA has rather powerful media features, which are built around the PJMEDIA conference bridge. I'm having a bit of trouble using the conference bridge with udp streams using the pjmedia API - I'd appreciate it if anyone can point me in the right direction. It's more clear now. Please see… Mar 3, 2016 · How much simultaneous calls can pjsip with pjmedia using a bridge for each call handle? Thx On 3/2/2016 5:10 PM, Bill Gardner wrote: > Hi Alaa, > > It's pretty easy to code an announcement system using pjsua, and if > you don't want to use pjsua you can look at the pjsua code as a guide. As an example, consider the following output: Apr 25, 2025 · Conference Bridge Relevant source files This document describes the Conference Bridge component of PJMEDIA, which provides a powerful and efficient mechanism for routing audio flows between multiple ports and mixing audio signals when required. a call) can transmit to multiple destinations, and one destination can receive from multiple sources. 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. Each call is add to default conference bridge in method: pjsua_media_channel_update () and then when i want add call to my conference bridge i have to create port add it to my conference bridge and remove call from default conference bridgefor this i need internal structure pjsua (pjsua_internal. I cant find any good code for make conferance. IMHO this is not clean The conference bridge calls pjmedia_port_get_frame() for all ports in the conference bridge, then it mixes the signal together according to ports connection in the bridge, and deliver the mixed signal by calling pjmedia_port_put_frame() for all ports in the bridge according to their connection. My log shows that I have Alsa linked in and that Alsa can find the desired device: 11:05:51. The ToC below shows the topics covered by this guide. The conference uses the signal level of each sources as a parameter to the mixing algorithm. The strong point for pjsip would be in my opinion pjmedia conference bridge (easy to use and powerful mixing, volume control, audio format conversion), this basically has no equivalent in baresip. ConfBridge (Asterisk 11+) - This is a conference bridge application based that supports wide band mixing. > Basically when the call connects you create a wav May 14, 2008 · Ok thank you for the explanation. The delay is caused by buffering to accommodate the audi Dec 14, 2023 · Have implemented pjsua with third party media and own audio device for an embedded product without conference bridge. - That would require support of multiple channels from PortAudio to conference bridge (todo) Jack client x output port 1 - conferenc bridge - rx transport stream 1 Jack client x output port 2 - conferenc bridge - rx transport stream 2 Jack client x output port 3 - conferenc bridge - rx transport stream 3 . conf files. 3k次，点赞19次，收藏18次。在分析问题前，我们先要对会议桥 (conference bridge)有个基本认识。从上图可以看出，会议桥起到一个承上启下的作用，是媒体设备与媒体流之间的交换重点。很显然，从设备到会议桥有个 sound device port 的对于 pjmeida_port 的实现，这里需要我们关注下。有了 Jun 5, 2018 · 文章浏览阅读3. Sections are identified by names in square brackets. I Now: I cannot find, in documentation or example/source code, how to connect a hardware device to the conference bridge. The video conference bridge shares the same principles as the audio conference bridge; application connects video source to video destination to allow video flow from that source to the specified destination, which in turn may also induce video mixing and duplicating operations. However Mar 23, 2018 · 文章浏览阅读5. This describes the conference bridge implementation in PJMEDIA. Any other participants connected via chan_pjsip will get the From display name, content-type, and body. Added sound device default:CARD Abstract: This paper presents our design of launching WebRTC and PJSIP in an IP-based Private Branch Exchange instance running in OpenStack private cloud. org/repos/wiki/FAQ#high-perf for creating multiple conference bridges running at different clock rates. Any idea where I should check next? dial-peer on Cisco dial-peer voice 11111 voip description Conference Dial Peer Testing destination-pattern 9182 redirect ip2ip session protocol sipv2 session target ipv4:10. Sample to mix multiple files in the conference bridge and play the result to sound device. 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. conf is a flat text file composed of sections like most configuration files used with Asterisk. Hi Thanks for reply. 1 Bug fixes SIP authentication: #4195 Asynchronous conference bridge: #4198, #4200, #4208 Dialog event subscription: #4214 See also PJSIP version 2. PJSIP project. 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 Group PJMEDIA_CONF ¶ group PJMEDIA_CONF Audio conference bridge implementation. The principle is very simple, that is you connect audio source to audio destination, and the bridge will make the audio flows from the source to destination, and that’s it. This file is pjsip-apps/src/samples/confsample. 15. As an example, consider the following output: Conference port options. Do not only connect microphone to conference bridge. Use pjsua’s cl (conference list) command from the pjsua ’s menu to check if the connection is made between the call and the sound device in the conference bridge. May 22, 2025 · Core Libraries Architecture Relevant source files Purpose and Scope This document explains the layered architecture of PJSIP's core libraries and their interdependencies. here. With the conference bridge, each conference slot (e. Jun 30, 2024 · 文章浏览阅读1. * This describes the conference bridge implementation in PJMEDIA. The conference bridge provides powerful and efficient mechanism to route the video flow and combine multiple video data from multiple video sources. (see SectionName below) Each section has one or more Specify maximum number of media ports to be created in the conference bridge. Creating a bridge for each call is more complicated but will allow your server to take full advantage of multiple cores and hence handle more calls it to the conference bridge master port. c The conference bridge The conference bridge provides a simple but yet powerful concept to manage audio flow between the audio medias. (I can talk and the 2 other destinations that I call can listen me and I can hear them). Conf connect: 1 --> 0 It seems like the call is not getting connected to the conference bridge properly, but we dont know why. When I connect a wav_player to a stream, the audio comes through okay - only when I deinitialise I get a segfault. The main benefits of using the switchboard are its ability to handle encoded audio frames, its low latency, and higher performance. id) error: Invalid value or argument Nov 15, 2019 · How to implement Conference calling with pjsip android? I can put my current call on hold and un-hold it successfully. Thank you for all this work. PJSUA Command Line Interface (CLI) Manual Table of Contents PJSUA Command Line Interface (CLI) Manual Introduction Commands Root commands Call and related commands [call] IM and Presence commands [im] Account commands [acc] Conference and Media commands [audio] Status and config commands [stat] Video commands [video] Introduction CLI is a feature of pjsua that enables user to execute commands The Audio Conference Bridge ¶ The conference bridge provides a simple but yet powerful concept to manage audio flow between the audio medias. John Smith. org) 1. The focus is on understanding the architectural design, dependency relationships, and how Oct 15, 2019 · The video conf bridge needs to know the resolution of all ports, so if any port updates the resolution (e. Attach rec port to conference bridge and do some routing with pjmedia_conf_connect_port. It covers the seven main libraries that comprise the PJSIP stack: PJLIB, PJLIB-UTIL, PJSIP, PJMEDIA, PJNATH, PJSUA-LIB, and PJSUA2. For supported channel drivers (currently only PJSIP) it’s now possible to convey multiple streams of media for multiple media types. 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 The videos compares the CPU usage between jitsi videobridge and pjsip conference. It seemed to be a tough nut for me. a Voice over IP/VoIP softphones). May 22, 2025 · The video media system handles camera preview, video calls, video windows, and multi-party video conferencing through a centralized video conference bridge architecture. . I am implementing Audio Conference Call in Android using Pjsip library v2. Each section defines configuration for a configuration object within res_pjsip or an associated module. Since all media terminate in the bridge (calls, file player, file recorder, etc), the value must be large enough to support all of them. Nov 24, 2023 · pjmedia_conf_add_port， 这个函数的目的是： Add stream port to the conference bridge 从pjmedia_conf_add_port函数的实现来看，入口参数strm_port必须是已经存在的pjmedia_port实例 源码提供了一个通过会议桥混音的例子confbench. Is this possible? If so, which type of port should I use for this purpose? I have tried various combinations of the memory capture / playback ports, but without success; when I read from the bridge master port, all I get are zeroed frames. Jun 5, 2018 · pjsip conference分析_宝贝等等我的博客-爱代码爱编程 2019-07-03 分类: pjsip conference pjsip confer Pjsip的Conference会议桥，主要的功能是抽象media的输入输出为port，并把port中的PCM数据进行混音，已达到多方通话的混音功能。 Hi, I was wondering if there is a means by which I can throw PCM frames straight into a conference bridge. > > Regards Switchboard Audio switchboard is drop-in (compile-time) replacement for the Conference Bridge. Open the source file for more information. The conference bridge acts as a central hub for audio communication, allowing multiple audio sources and destinations to be connected in flexible Conference port options. 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. The major contributions were the following: (1) design how to integrate open-source software devices working with WebRTC in a Performance Optimization Maximising performance Echo canceller Float vs fixed point Codec Avoid resampling Choose effective sampling rate Conference bridge vs audio switchboard Logging Threads Run-time checks Stack checks Safe module Unescape in-place Hash tolower Optimization Release mode How to configure pjsip to serve thousands of calls Conference Bridge Bidirectional Port Echo Cancellation Port Buffer Playback Capture to Buffer Null Port Resampling Port Multi-frequency/DTMF Tone Generator Audio Stream Video Stream WAV File Playback WAV Playlist WAV File Recorder Media channel splitter/combiner Clock provider Sep 24, 2014 · Im working on the sip android app. Mar 5, 2019 · The video mixing operation is done by video conference bridge, which will use very similar APIs to the audio conference bridge. Then,what is the use of creating a conference bridge and adding ports to it to conference calls. Your debug message show because, if you want to using conference bridge, you have to connect it to a sound device with following function (pjmedia_snd_port_create or pjmedia_snd_port_create_player). It implements the Session Initiation Protocol (SIP), media handling capabilities, and various network protocols to facilitate the development of VoIP applications, video communications, and instant messaging. 16 is released! Real time text (RTT) (RFC 4103) Parallel conference bridge Audio video synchronization Download Group PJSUA_LIB_MEDIA group PJSUA_LIB_MEDIA Media manipulation. 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. May 22, 2025 · The conference bridge (pjmedia_conf) is the central component that manages audio flow between all audio media objects. The conference bridge has 120ms (buffer size 6 * ptime 20 ms) latency. the media port. Check audio interconnection in the conference bridge Use pjsua’s cl (conference list) command from the pjsua ’s menu to check if the connection is made between the call and the sound device in the conference bridge. When bridge_softmix (the bridging module used by ConfBridge) sees the message, it relays it to all other bridge participants. 4k次。本文深入解析了会议桥的实现机制，包括其核心组件如master_port和snd_dev_port的作用，以及如何通过pjmedia_snd_port创建声卡设备接口。此外，还探讨了会议桥中的passive port与外部pjmedia_port的工作原理。 The video conference bridge shares the same principles as the audio conference bridge; application connects video source to video destination to allow video flow from that source to the specified destination, which in turn may also induce video mixing and duplicating operations. quality setting, which default value is PJSUA_DEFAULT_CODEC_QUALITY indeed. c . Anybody pls do help with conferencing. Ad-hoc Multiparty Bridges (Asterisk 12+) - Some DTMF features like 3-way attended transfers can create multiparty bridges as necessary. 849 alsa_dev. Am I doing something completely wrong here? Is there a default approach on how to connect more than 2 IP-telephones with pjsip? I couldn't find anything like this in the examples directory. Can someone help me? Thanks. Try to isolate Call/Account code creating custom classes and call from different threads. Configured the dial-peer and dialing the conferencing number no luck getting through. The conference bridge provides powerful and very efficient mechanism to route the audio flow and mix the audio signal when required. The run-time option when creating the resample port or conference bridge should be used instead. Always Be Conferencing Automatically join friends to your Conference Bridge before they even answer the phone! Compatible with ASTERISK versions 13+, using AEL-based dial plan configuration file, that you can access from your existing extensions. Thanks in advance. 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. But again, this only applies if you use PJSUA-LIB. It sounds like a gentle hissing or static sound, and it's constant regardless of whether people are talking or not. 248 dtmf-relay rtp-nte codec Group PJMEDIA_VID_CONF group PJMEDIA_VID_CONF Video conference bridge implementation destination. It implements standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. As an example, consider the following output: Nov 25, 2025 · PJSIP version 2. So I'm pretty sure the problem is with how I've tried to follow the pjsua-lib-perf FAQ. After inv Nov 25, 2014 · I am setting up asterisk as a conference number on Cisco 2800 CME. If a pjsip callback is busy (you use it until a long task end) other callbacks won't trigger so it is impossible to make it work. Its main drawback is it doesn’t do conferencing. There are several types of video media objects supported in PJSUA2: Use pjsua’s cl (conference list) command from the pjsua ’s menu to check if the connection is made between the call and the sound device in the conference bridge. g. As an example, consider the following output: PJMEDIA Samples Below are PJMEDIA samples. This is strange as I haven't connected any media port to the conference object at this time. ***I can't see any issues in the log file, and I've confirmed the key 'on_' callbacks all definitely run* - The log file prints out the following - which to me looks May 22, 2025 · PJSIP is an open source multimedia communication library written in C that implements SIP (Session Initiation Protocol) and related protocols for voice, video, and instant messaging applications. g 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, embedded Android Getting Started: Building Android SIP VoIP and Video Client Application This guide provides step-by-step instructions to build sample Open Source Android SIP VoIP and video client applications using PJSIP, a powerful, small footprint, and portable multimedia communication library. 2015-09-10 08:15:08: @nanangizz created the issue on trac ticket 1884 Reported that in an audio call session, after disconnecting audio flow from microphone to call stream in conference bridge, remote hears audio stutter badly. But my goal is to make a conference call which everyone can talk and listen at the same time. In video call using H264 codec, the exact resolution of incoming video will only be known a wav_writer and a wav_player working no problem with the default pjsua conference bridge. 3. If you use PJSUA-LIB, this is controlled by pjsua_media_config. Jitsi videobridge uses 80% and pjsip uses 47%, so pjsip is the winner. Conference Bridge Bidirectional Port Echo Cancellation Port Buffer Playback Capture to Buffer Null Port Resampling Port Multi-frequency/DTMF Tone Generator Audio Stream Video Stream WAV File Playback WAV Playlist WAV File Recorder Media channel splitter/combiner Clock provider PJSIP Configuration Sections and Relationships Configuration Section Format pjsip. Disk files (to players) and network connections (to recorders) works fine. 100. Regards 0 Replies 2 Views Permalink to this page Disable GitHub Gist: instantly share code, notes, and snippets. (for e. The values here can be combined in bitmask to be specified when the conference bridge is created. It provides a simple but powerful model where audio sources connect to destinations, with automatic mixing and duplication. kymi kgelg jbw cuvtbr vuklsso vkslkroe ltyum nffprer mxuku ydotv