Pjsua multiple calls.
Group PJSUA_LIB_ACC group PJSUA_LIB_ACC.
Pjsua multiple calls Multiple backends: The new API supports multiple audio backends (called factories or drivers in the code) to be active simultaneously, and audio backends may be added or removed during run-time. Stack Overflow for Teams Where developers & technologists share private knowledge with coworkers; Advertising & Talent Reach devs & technologists worldwide about your product, service or employer brand; OverflowAI GenAI features for Teams; OverflowAPI Train & fine-tune LLMs; Labs The future of collective knowledge sharing; About the company The seemingly simple solution is to modify the pjsua source. This issue seems not only to be a PJSUA/PJSIP issue, but a platform specific issue (Mac OS X) as the (nearly) identical steps were performed on Ubuntu 15. I want to call 4444 from a sip phone Eg: Twinkle calling 4444 running on pjsua 4. PJSUA2 is a C++ library, which you can find under pjsip directory in the PJSIP distribution. I create pjsua modules in the main thread with APIs such as pjsua_create and pjsua_init, then I call pjsua_acc_add or pjsua_call_make_call in another thread, it usually crash. Am getting PJSUA_core. I know, we can start transmitting various ports using startTransmit() in Audio Media. Features: Command completion, the system will detect if a fraction of a word makes up a unique command. Overview on using the API Getting started Configure the application’s project settings. If my memory serves me right, SDP is not processed until a 文章浏览阅读873次。本文介绍如何通过修改pjlib配置文件和pjsua核心代码，将最大并发调用数从4增加到32的过程。这涉及到对pjsua_max_calls宏的重新定义及相应函数的调整，确保软件能同时处理更多呼叫。 You still need to specify your SIP URI with the –id option to identify yourself in calls. Understanding the Configuration. Specify how support for reliable provisional response (100rel/ PRACK) should After continuous attempts I solved it myself. enumerator PJSUA_STATE_RUNNING After pjsua_start() is called and before pjsua_destroy() is called. thread_name – Mutex Locks Order in PJSUA-LIB. getNatConfig(). org Subject: [pjsip] Multiple calls in pjsip Hi, 1. pjsip-perf. Information of the transport id. c !Sending 2 bytes keep-alive packet for acc 0 to 10. In SIP terms, the identity is used as the From header in outgoing requests. Maximum number of calls configured. Public Functions. This docker image support audio, if sound For using video with PJSUA-LIB, see Video User’s Guide. But, firstly, how to initiate a conference call? anybody pls help. After pj_init() is completed, application can continue with the initialization or create a secondary/worker thread and register the thread by calling pj_thread_register(). We have done following implementation for handling multiple calls : After this callback is called, normally PJSUA-API will disconnect old_call_id and establish new_call_id. After this I would expect the call goes from PJSIP_INV_STATE_CONNECTING to PJSIP_INV_STATE_CONFIRMED, but it does not happen, so PJSIP continues to send a 200 OK and receive the ACK every about 2 seconds, until the call times out after 32 seconds and PJSIP disconnects the call (sending a BYE). Media manipulation. There are many types of mutexes used in PJSIP, and both the library and application MUST obey a uniform lock order to avoid deadlocks. Hi all, I need to do bulk call testing with my gateway. 070 pjsua _acc. This can be Explanation of changes: PJSUA_MAX_CALLS: Adjusts the maximum number of simultaneous calls. Hangup active calls or continue the call by sending re-INVITE. > For example, if you make a call, the result of that call is capturate in one > thread that manage all the calls of the system. But, could not record calls. h file?. huebner@xxxxxxxxx> To: "pjsip list" <pjsip at lists. App can use on_dtmf_digit2() to monitor incoming DTMF using the method in pjsua_dtmf_method. sipecho. 0. The mentioned line in pjsua_core. Account configurations . I have read in "PJSUA2 Book" (3 PJSUA2-High Level API / General Transaction tsx0x7f3a54012278 state changed to Calling 12:08:36. PJSIP-UA Show or Hide Incoming Video . You can either hangup or maintain the ongoing/active calls. API Reference Audio Device API After pjsua_init() is called but before pjsua_start() is called. c Sending When PJSUA sends second INVITE request right after first OK was received, which it does in in-dialog mode (you can check that with wireshark), onCallMediaState is called second time and, if for some reason parties agreed on some another codec, not that that was the first time, there could be another getMedia(i) object, and the first original Introduction to PJSUA2 . pygui. h" PjsuaThread *pjsua_thread; MainWindow::MainWindow(QWidget *parent) : QMainWindow(parent) , ui(new Ui::MainWindow) { ui->setupUi (this When calling multiple C++ functions from QML and the first function contains a connect call, the QML continues before the callback is triggerd. Endpoint ¶. To see the full help for it, see "core show application Dial" on the Asterisk CLI, or see Dial. @Override PJSIP_DIALOG_CAP_SUPPORTED if the specified capability is explicitly supported, see pjsip_dialog_cap_status for more info. PJSIP_TRANSPORT_UDP, sipTpConfig) # Start the library ep. I'm trying to do a conference call between 2 asterisk extension, managed by pjsip. This will also affect ICE completion update in updating default address in SDP. References: pjsua_transport_config. Note that on_dtmf_digit() will not be called once Sample. Calls, presence, and buddy will be explained in later chapters. process ID. create(accountConfig) method when logging in. enumerator PJSUA_STATE_STARTING After pjsua_start() is called but before everything is running. More settings can be specified in pj::AccountConfig:. Hold/Unhold current call. This answer is combination of code snippets from these two links: PJSUA-API Media Manipulation and pjsipDll_PlayWav. If there are multiple video streams in a call, the default video is the first active video media in the call. Application can then inspect this data by calling getUserData(). I try to use PJSUA-LIB to develop a server like application, but performance is poor. Application must call this function before calling any other functions, to make sure that the underlying libraries are properly initialized. Despite modifying all relevant flags in config_site. Among other things, it supports: multiple accounts, with each having different registration settings, multiple SIP calls (with or without conferencing), SIMPLE/presence with SUBSCRIBE/NOTIFY, PUBLISH, support for PIDF and X-PIDF; instant messaging with MESSAGE request, call Functions. Parameters:. You wanna look into pjsua_call_make_call(), pjsua_call_answer() and pjsua_call_get_media_session(). count – In input, specifies the maximum number of elements The issue is that I am unable to make videocalls via SIP work. Suppose we want to talk with two remote Subject: Re: [pjsip] pjsua and multiple calls. pjsua_100rel_use prackUse. void libCreate PJSUA2_THROW(Error). 10) and make everything work on Pi Zero 2W. In order to receive call in the other account , you need to traverse the My original question related to the pjsua_set_snd_dev () call that operates on the global pjsua_var variable. 0. Assuming you have the Account object as acc variable and destination URI string in dest_uri, you can initiate CLI is a feature of pjsua that enables user to execute commands from telnet/console interface. accountConfigurationg. Group PJSUA_LIB_MEDIA group PJSUA_LIB_MEDIA. Using the Audio Device API See Using Audio Device API. According to SIP spec, a request is sent to the address in the destination URI, which is the URI in the Route header if it is present, or to the request URI if there is no Route header. for click-to-call. In pjsip-pjsua during the account configuration we set . libStart() # Account configuration acfg = pj. I've pjsua_set_snd_dev in NULL. unsigned pjsua_call_get_count (void) ¶ Get number of currently active calls. pj::AccountCallConfig, the call settings, such as whether reliable provisional response (SIP Get maximum number of calls configured in pjsua. The IP change progress operation. exe from source for Windows using the Win32 Release configuration. Application may then query the call info to get the detail call states. We are assuming you already know a little bit about the Dial application here. Adjustments to increase the number of calls involve understanding both the settings within your codebase and the implications they have. libInit(ep_config) # Whatever ep_config you're using ep. Ticket #1371 reports another soft deadlock case when working with multiple calls. Application may then query the information and state of each call by calling pjsua_call_get_info(). I want to call 2222 from a sip phone Eg: Ekiga calling 2222 running on pjsua 3. setNullDev() 我们在PJSIP和Asterisk上的max_calls设置有问题。我们正在对我们的Asterisk服务器进行压力测试，但发现我们的PJSIP模块最多有32个活动呼叫限制。我们使用PJSIP来测试我们的Asterisk服务器在快速搜索之后，我们发现以下设置可以解决这个问题。 Following steps can be taken to increase number of calls suppo Public Members. PJSUA accounts provide identity (or identities) of the user who is currently using the application. pj::AccountRegConfig, the registration settings, such as registrar server and retry interval. multiple calls using pjsip Padmaja 2009-07-23 05:10:36 UTC. net> Hi, I was trying to configure Pjsua to run above 32 simultaneous calls by changing Crash after calling PJLIB APIs using Grand Central Dispatch (GCD) Audio lost or other issues with interruption (by a phone call or an alarm), headset plug/unplug, or Bluetooth input; SIP transport keepalive while in background; PJSUA-LIB PJSUA-LIB is a library that integrates PJSIP, PJMEDIA, and PJNATH into high-level, easy to use API for CLI is a feature of pjsua that enables user to execute commands from telnet/console interface. Get library version. a Voice over IP/VoIP softphones). pool – The memory pool from which the thread record will be allocated from. pjsua_ip_change_op op ¶. So, here is the code: #include <iostream& we can hold and accept calls (max 2 calls). pj_status_t pj_thread_create (pj_pool_t * pool, const char * thread_name, pj_thread_proc * proc, void * arg, pj_size_t stack_size, unsigned flags, pj_thread_t * * thread) . Supports UDP, TCP, IPv6. h" #include "pjsuathread. c bind error: Address already in use when I want to make multiple calls at the same time in the console app How can I make multiple calls at the same time. on_call_transfer_request Crash after calling PJLIB APIs using Grand Central Dispatch (GCD) Audio lost or other issues with interruption (by a phone call or an alarm), headset plug/unplug, or Bluetooth input List of supported SIP features and link to the relevant PJSIP I used PJSIP Library and able to register user with server but i cannot make a call is it possible to make call without sip-client or to make to itself ,, cannot make calls with server Sample. The null I'm trying to develop a code in Python that first makes a sip call to an extension and when the call is answered it plays an audio file, I managed to authenticate the account but the call is not made, below is my code, me too I want to implement call reception in sequence, I can even see the call arriving but I don't know how to configure it to All SIP features supported by PJSIP should be available from PJSUA. but after sucesfull setup, I will recompile the PJSIP (version 2. For example, if you make a call, the result of that call is capturate in one thread that manage all the calls of the system. newCall. I am not sure why it even tries, because it should use UDP by default anyway Hope this helps! DISCLAIMER: I know this is an old post, and that it probably shouldn't belong here, but I have been searching for the same answer for some time, and there are not many resources out there. The class pj:: And if there are two or more concurrent video calls sharing the same capture device, the device will be transmitting to three or more destinations. Set pjsua to use null sound device. 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. If you intend to maintain the Group PJSUA_LIB_ACC group PJSUA_LIB_ACC. pj_status_t status ¶. invtester. However, when I attempt to answer a call (using 180 or 200), like so: 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. I had tried pjsua's AudioMediaRecorder but since there in no proper documentation I did not understood anything. Last modified 14 years ago Last modified on Sep 16, 2011 2:55:01 AM. com”). Is it possible to have the following callback Making outgoing call is simple, just invoke makeCall () method of the Call object. 10 install:. 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 ref: pjsua_acc_config::allow_contact_rewrite and pjsua_acc_config::contact_rewrite_method. That is quite expected, unfortunately. Post by Peter Lukac Hello all, I have 2 audio devices and i want these devices use for calling at the same time. Download in other This method will call pjsua_call_dial_dtmf() when sending DTMF using PJSUA_DTMF_METHOD_RFC2833. pj_status_t pjsua_enum_calls (pjsua_call_id ids [], unsigned * count) ¶ Enumerate all active calls. How to implement Conference calling with pjsip android? I can put my current call on hold and un-hold it successfully. Bye. Any IncomingSubscribeParam. 10 and it was able to create additional accounts with no issue. Subsequent operations to the call can use the method in the call instance, and events to the call will be reported to the callback. /ui_mainwindow. I Calls are represented by pj::Call class. Make outgoing call is by invoking pj::Call::makeCall() with the destination URI string (something like “sip:alice@example. The SIP and media features and object modelling follows what PJSUA-LIB provides (for example, we still have accounts, call, buddy, and so on), but the API to access them is different. Version libVersion const ¶. com. Issue : The issues we are facing are : We are able to accept second call which have no audio when hold and accept. More on the callback will be explained a bit later. If I understand what you write, you are multiple threads within one process to handle Hi, I am trying to run Pjsua with 10-20 simultaneous calls (5 Calls per second) and having both the --id (user name part) and the destination (also user name part) increased in every session. Note that this value should only be used to compare multiple events received via the same method relatively to PJSUA (project page) is an CLI and curses SIP softphone, part of the PJSIP stack. Utility to send INVITE or re-INVITE without SDP, for testing. org] On Behalf Of Benny Prijono Sent: Thursday, March 19, 2009 3:15 PM To: pjsip list Subject: Re: Pjsua with more than 32 calls 2009/3/18 Noga Yehudai <nogab at juniper. Describe the bug I successfully built the pjsua/pjsip. Making outgoing calls . void setUserData (Token user_data) Attach application specific data to the call. libCreate() ep. In pjsua the last account wiil be considered as current account, it can only receive the incoming call to this account. To use PJSIP, it is recommended to call pj_init() and pj_shutdown() from the main thread. Consistent across different files ensures stability across call management. The developer must change the Call-ID between calls creating a unique id for each outgoing call. Get process ID. To increase this limit, the library must be recompiled with new PJSUA_MAX_CALLS value. pjsip. The value specified here must be smaller than the compile time maximum settings PJSUA_MAX_CALLS, which by default is 32. Default: PJSUA_CALL_HOLD_TYPE_DEFAULT . PJSUA Accounts management. #include ". Regardless of the setting above, you can use the following steps to show or hide the display incoming video: Use pjsua_call_get_vid_stream_idx() or enumerate the call’s media stream to find the media index of the default video. PJSUA2 supports more than one simultaneous calls. audDevManager(). Returns. See pj::Call class for more info. I want to call 123 and 124, and make them to talk. More info can be found here. setIceEnabled(true); So, many datas are getting passed to the server like rtcp payloads etc. Virtual destructor . Creating a secondary thread is especially recommended, As a convention in PJSUA-LIB API, port zero of the conference bridge is denoted for the sound device. 21:5060 12:08:36. When increasing the limit, compile time options PJSUA_MAX_CALLS and PJ_IOQUEUE_MAX_HANDLES also needs to be changed accordingly (set the later to approximately 3 times PJSUA_MAX_CALLS). h, the maximum number of active calls I can achieve with a singl I need to manage multiple calls simultaneously. Permalink. c: cfg->max_calls = ( ( Maximum simultaneous calls. /configure CFLAGS=-fPIC CXXFLAGS=-fPIC sudo make dep && make sudo make cd pjsip MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. Anybody pls do help with conferencing. k. I sucessfully connected audio calls, and even other requested features like "unlocking doors" (switching relay on GPIO when right DTMG code is entered) works. PJSIP only sends the request with TCP if the destination URI contains Using thread with PJSUA initialization and shutdown . Thanks to the video conference bridge for its duplicating feature. with state PJSIP_EVSUB_STATE_PENDING and later calls presNotify() again to accept or reject the subscription request. Is it possible to create a Number of current calls. c simply compares and assigns the value of PJSUA_MAX_CALLS with a hard limit of 400. Video media is similar to audio media in many ways. pj_status_t pjsua_enum_calls (pjsua_call_id ids [], unsigned * count) Enumerate all active calls. To use the Call class, application SHOULD subclass it, such as: Application implement Call’s callbacks to process events related to the call, such as I've noticed if you are handling an incoming call on one account any incoming calls on other accounts that occur at same time are delayed. pjsua --id="sip:user1@ip_address" Make a call to the SIP URI to and you should receive a call on the terminal. PJSUA-LIB API itself is a library that unifies SIP, audio/video media, NAT traversal, and client media application best practices into a high I am new to PJSUA2, and I'm trying to make calls using this library. Library(s) Description. Noga _____ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists. If application wishes to stop the playback, it can stop the media transmission in the callback, and only after all transmissions have been stopped, could the application safely destroy the player. we can switch between calls. ar> wrote: > yes, un thread per call, but PJSUA driver all the events in only one thread. Command history (the use of up and down arrow). c. I don't know your specific case but in case your machine lacks sound device you need to set the following in your endpoint: ep = pj. PJSUA is an invaluable tool Multiple backends: The new API supports multiple audio backends (called factories or drivers in the code) to be active simultaneously, and audio backends may be added or removed during run-time. After a great effort, I compiled and integrated pjsip/pjsua2 into my app with which client registrations and calls are working fine. Hence connecting a media to port zero will play that media to speaker, and connecting port zero to a media will capture audio from the microphone. I've set up two different transports and two accounts, this i Group PJSUA_LIB_ACC group PJSUA_LIB_ACC. newCallId, and return the Call object via prm. To be able to control the call, e. A pjmedia_port_get_frame() call by conference bridge to media stream (pjmedia_stream) will cause it to pick one frame from the jitter buffer, decode the frame using PJSIP libraries provide multi-level APIs to do SIP calls, presence, and instant messaging, as well as handling media and NAT traversal. It allowing to do The simplest became the hardest. AccountConfig() acfg Subject: [pjsip] using pjsua with multiple calls Hi, I am trying to run Pjsua with 10-20 simultaneous calls (5 Calls per second) and having both the --id (user name part) and the destination (also user name part) increased in every session. I want create call and connect it to first device (mic/speaker) and But when I use it in my android project (though JNI calls), it usually crash when I register user to sip sever or make calls out. If the file or playlist is set to play repeatedly, then the callback will be called multiple times. Post by Hans Hübner. Explore how to modify maximum call limits in PJSUA2 within an iOS Swift Xcode project, emphasizing changes in configuration files for optimal performance. on_call_state Notify application when invite state has changed. code other than 200 and 202 will be treated as 200. I've simplified my script as much as possible to replicate the issue below. Useful for auto-responding test server. I registered two numbers/accounts with my pjsua application (Eg: 2222, 4444) 2. PJSUA2 wraps together the signaling, media, and NAT traversal functionality into easy to use call control API, account management, buddy list If you are using PJSUA-LIB, then the maximum number of calls supported is configurable from pjsua_config::max_calls (default is 4). Steps completed on working Ubuntu 15. I'm unable to receive calls on my PJSUA2 python script. Jose ----- Original Message ----- From: "Hans H?bner" <hans. . It seemed to be a tough nut for me. Is it enough to change only this line to modify the maximum number of calls in PJSUA2? pjsip\src\pjsua-lib\pjsua_core. PJSUA has rather powerful media features, which are built around the PJMEDIA conference bridge. pjsua2_demo. org> Sent: Monday, August 22, 2011 5:56 PM Subject: Re: pjsua and multiple calls > On Mon, Aug 22, 2011 at 10:42 PM, Jose Suarez <jsuarez at padirac. PJSUA2. Create a new thread. Dialing from dialplan¶. e. Parameters: ids – Array of account IDs to be initialized. 8 with pjsua2 using swig (built using the files given in the pjsip source code) I can make calls fine. The key is to create a pjmedia_session out of your custom SDP on code paths of both incoming and outgoing calls. g: hold, transfer, change media parameters, application must instantiate a new Call object for the new call using call ID specified in prm. Demonstrates basic usages of PJSUA2. pj::AccountSipConfig, the SIP settings, such as credential information and proxy server. The operation progress status. I already managed to authorize on remote sip server, but calling is more difficult. If I keep on giving wrong credentials, same But in general, my tip would be that the mailing list should be your goto place for asking PJSIP/PJSUA questions instead of StackOverflow as there is probably more people reading it who have been On Mon, Aug 22, 2011 at 10:21 PM, Jose Suarez <jsuarez at padirac. CLI mode is enabled/disabled by running pjsua with these options: sip_uri = (pjsip_sip_uri*) pjsip_uri_get_uri(call->inv->dlg->target); Subject: pjsua and multiple calls; From: jsuarez@xxxxxxxxxxxxxx (Jose Suarez) Date: Mon, 22 Aug 2011 17:35:12 -0300; Yes, I'm not using audio devices, sorry, I don't figure that in your case. enumerator PJSUA_STATE_CLOSING After pjsua_destroy() is called but before the There is one Account instance and I'm calling account. This tutorial uses PJSUA-API, the highest layer of abstraction of all, which combines PJSIP (the SIP stack library) and PJMEDIA (the media stack library). 070 pjsua_acc. However, the point it crashed on is quite Please see the reference documentation for Account for more info. Parameters: user_data – Arbitrary data to be attached to the call. Default constructor . I could find the To: ***@lists. I'm making calls to a voip provider. Bitmask of pjsua_call_flag constants. Number of currently active calls. Note that on_dtmf_digit() callback can only monitor incoming DTMF using RFC 2833. 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. Endpoint() ep. pj_uint32_t pj_getpid (void) . Instantiate pjsua application. pjsua_transport_config_default() pjsua_transport_create() Sending Initial Requests . PJSUA2 API is the highest API from PJSIP, on top of PJSUA-LIB API. PJSUA-API supports creating and managing multiple accounts. This simplified script is largely sourced from the PJSUA2 docs (pj. I have a question to my ios-swift-pjsua2 Xcode project. A Simple SIP User Agent. ar> > wrote: >> I'm thinking to rewrite all my code using PJLIB The snippet above initiates outgoing call to dst_uri, install the callback to the call, and saves the call instance in call object. Switch call button from call kit is disabled. Application may then query the I know the only solution is using the NDK. Note that an account instance is required to create a call Or should other configurations be added to the config_site. CLI mode is enabled/disabled by running pjsua with these PJSIP Tutorial (Using PJSUA-API) As you can see from the diagram in PJSIP Documentation page, PJSIP software consists of multiple API abstractions. PJSIP-UA. For that I need to send 4 did multiple call testing using pjsip, please advice. Returns:. These features will be described later in this chapter. Please see the documentation on pjsua_call_hold_type for more info. After this callback is called, normally PJSUA-API will disconnect this call and establish a new call. Thanks, Padmaja Joel Dodson 2009-07-23 05:28:45 UTC. Application implement Call’s callbacks to process events related to the call, such as pj::Call::onCallState(), and many more. TransportId transportId ¶. Post by Jose Suarez yes, un thread per call, but PJSUA driver all the events in only one thread. My application is an automatic dialer that I'm using in an Elastix Server. Add the following include: I am using Python 3. sudo . Specifies if lock codec feature should always use INVITE method. Within Linux, users could use it as a phone to dial numbers right from the command line, i. So, the library use the more reliable TCP connection automatically. PJSUA2 (Python) Python GUI application supporting audio calls, presence, and instant messaging. When pjsua makes a call it is using ports (conference ports) for transferring media from/to the call destination to your device speaker. Arguments/command-params completion. cpp. It facilitates high quality VoIP calls (p2p or on regular telephones) based on the open SIP protocol. PJSIP (core) Simple implementation of stateful proxy as spec-ed by RFC 3261. 070 tdta0x7f3a44005248 Destroying txdata raw 12:08:36. The C++ The solution for me was to add the option --no-tcp to the configuration, which avoids using TCP. Accepts all incoming calls with SDP to make caller send media to itself. virtual ~Endpoint ¶. Below we'll simply dial an endpoint using the chan_pjsip channel driver. iiyftcbufqwirazdhsysrqtbsxymjmeisjgmbaricotyastrkqtnmdoysklyhbqjqhsfkgwdh