iphone Live call Open source Framework Pjsip compile-pjsua run test

iphone real-time call in the open source framework, PJSIP is a relatively streamlined framework, more than Linphone good compile. The following steps are described in compiling the run.

First, compile the operating environment:

iphone:5.1.1, System: 10.7.3, xcode:4.5.2

Ii. Preparatory work:

1, from the PJSIP official website download tar source code, decompression to the local;

2, in the directory/pjsip/pjlib/include/pj/new config_site.h, paste the following code in:

#define Pj_config_iphone 1#include <pj/config_site_sample.h>

Third, compile Pjsip:

Run the following command in the terminal:

$ cd/path/to/your/pjsip/$./configure-iphone$ make dep && do clean && make

Iv. Compiling PJSUA Demo Project:

Open the item in the directory Pjproject/pjsip-apps/src/ipjsua, connect your iphone to compile and run it on the line.

Five, Test call:

1. VoIP Server:

You can use the Minisipserver free version, the installation is simple, after installation do not forget to create a new account (here are 100 and 101 for example).

2. VoIP Client:

Personal feeling x-lite is better, whether it's a Mac version or a Windows version. Successful login will show successful registration.

3. Login to account in the iphone's PJSIP Interface 101:

+ayour SIP URL: (Empty to cancel): Sip:[email Protected]url of the Registrar: (Empty to cancel): Sip:192.168.1.1auth Realm : (Empty to cancel): *auth Username: (Empty to cancel): 101Auth Password: (Empty to cancel): 123456

and 100 Call:

+benter Buddy ' s URI: (Empty to cancel): Sip:[email protected]mmake call:1

The phone operation can also be simplified:

Mmake Call:sip:[email protected]

You can also add related information to the configuration article:

Add content at (ALICE.CFG):

# This was a comment in the config file.--id sip:[email protected]--registrar sip:example.com--realm *--username Alice--pas Sword Secret

=================================

Introduction to other configuration file usage:

Usage: pjsua [options] [SIP URL Call] General options:--config-file=file read Configuration/parameters from file;--help display this help screen;--version display version information; logging options:--log-file= fname log file name (default is stderr),--log-level=n set log maximum level is N (0 (None) 6 (Tracking)) (default = 5);--app-log-level=n set log maximum level is stdout display (default = 4) --color use a variety of logs (default on Win32);--no-color disables colorful logs;--LIGHT-BG uses the color of black text on white (default is dark background); SIP account options:--use-ims Open the 3gpp/ims settings associated with this account,--use-srtp=n use SRTP? 0: Not used, 1: optional, 2: Forced use (default: 0);--srtp-secure=n SRTP need secure sip? 0: Not required, 1:tls mode, 2:sips (default: 1);--registrar=url set the Url;--id=url of the registration server set the local account Url--contact=url Optionally overwrite contact information--contact-params=s add S parameter to the specified contact URI--proxy=url optional Access Proxy URL--reg-timeout=sec registration interval (default)--realm =string Setting the domain--username=string setting the user name--password=string setting the password--publish sending Publish--use-100rel requires a reliable temporary response (100rel)-- Auto-update-nat=n N is 0 or either enable/disable SIP traversal behind symmetric nat (default 1)--next-cred add additional credentials SIP Account Control:--next-account add more account transfer options:--ipv6 Use ipv6--local-port=port port--ip-addr=ip IP address--bound-addr=ip bind port--no-tcp Disable TCP Transport--NO-UDP disable UDP transport--nameserver=ns Domain Name server--outbound=url set the URL of the global proxy server, you can specify multiple times--Stun-srv=name Set stun server host or domain name TLS option:--use-tls Enable TLS transport (default is not on)--tls-ca-file Specify TLS CA file (default = None)--tls-cert-file Specify the TLS certificate file (default = None)--tls-privkey-file Specify the TLS private key file (default = None)--tls-password Specify the TLS private key file password (default is None)--tls-verify-server Verify the server's certificate (default = no)--tls-verify-client verifies the client's certificate (default = not)--tls-neg-timeout Specifies a timeout (default value none)--tls-srv-name Specify the TLS server name as a multihomed server (optional) media options:--add-codec=name manually add codecs (all by default)--dis-codec=name Disable a codec--clock-rate=n Overlay meeting Bridge clock frequency--snd-clock-rate=n overlay audio device clock frequency--stereo audio device and conference bridge turn on stereo mode--null-audio use a null audio device--play-file=file Registering a WAV file in a meeting bridge--play-tone=format registers a tone with the conference bridge, the format is ' F1,f2,on,off ', where f1,f2 is frequency, on,off=on/off, and can be specified multiple times. --auto-play Auto Play file (call only)--auto-loop automatically loops incoming RTP to outgoing rtp--auto-conf automatically joins the meeting--rec-file=file audio file (the extension can make. wav or. MP3)--auto-rec automatic recording of calls--quality=n specified media quality (0-10, default 6)--ptime=msec Overwrite codec ptime in milliseconds--no-vad deactivate VAD scheme/ Silent Detector (VAD enabled by default)--ec-tail=msec set echo offset tail length (default.--ec-opt=opt) Select the Echo cancellation algorithm (0 = default, 1 = speex,2 = suppressed)--ilbc-mode=mode Set ILBC Speech codec mode (20 or 30, default is)--capture-dev=id audio capture Device ID (default =-1)--playback-dev=id audio playback Device ID (default =-1)--capture-lat=n Audio Capture Delay (MS, default = +)--playback-lat=n Audio playback delay (MS, default = 1)--snd-auto-close=n idle n seconds after automatically turning off the audio device specify n =-A (default) disable this feature. Specifies that instant off is not used when N = 0. --no-tones Disable hear sound--jb-max-size specify maximum jitter buffer (frame, default = 1) Media transfer option:--use-ice use ICE (default: Not used)--ice-no-host disable the Ice host candidate (default: NO)--ice-no-rtcp disable RTCP component (default: NO)--rtp-port=n RTP attempt port cardinality (default 4000)--rx-drop-pct=pct drop pct percent of Rx RTP (for PKT lost s IM, default:0)--tx-drop-pct=pct drop pct percent of TX RTP (for PKT lost Sim, default:0)--use-turn Enable turn relay Wit H ICE (default:no)--turn-srv turn server domain or host name--turn-tcp use TCP to connect to turn server (default: NO)--turn-user turn user name--turn-passwd Turn password Friend list (can be multiple):--add-buddy URL adds the specified URL to the user agent option in the Friends list:--auto-answer=code answer code (such as $) that automatically answers incoming calls--max-calls=n Maximum number of concurrent calls (default: 4, Max: 255)--thread-cnt=n Number of worker threads (default: 1)--duration=sec set maximum talk time (default: No Limit)--norefersub Disable event Subscription--use-compact-form minimum SIP message size--NO-FORCE-LR allows strict routing--accept-redirect=n to be used to specify how to handle call redirection responses (3XX) when transferring calls. 0: Reject, 1: auto (default), 2: Ask

Iphone Live call Open source Framework Pjsip compile-pjsua run test