QT开发pjsip的VOIP,A8平台运行

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lishaman

7991人浏览 · 2013-08-01 15:54:43

lishaman · 2013-08-01 15:54:43 发布

QT开发pjsip的VOIP

开发环境

平台：A8

环境：Linux-3.0.8

实现功能：使用QT开发VOIP进行初始化、拨号、挂起

测试工具：minisipserver服务器

效果

界面：

minisipserver显示：

封装SIP操作接口：

#include "qt_sip.h"

qt_sip::qt_sip()

{

}

初始化

bool qt_sip::init_sip(void)

{

pj_status_t status;

/* Create pjsua first! */

status = pjsua_create();

if (status != PJ_SUCCESS)

return -1;

/* Init pjsua */

{

pjsua_config cfg;

pjsua_logging_config log_cfg;

pjsua_config_default(&cfg);

cfg.cb.on_incoming_call = &on_incoming_call;

cfg.cb.on_call_media_state = &on_call_media_state;

cfg.cb.on_call_state = &on_call_state;

pjsua_logging_config_default(&log_cfg);

log_cfg.console_level = 4;

status = pjsua_init(&cfg, &log_cfg, NULL);

if (status != PJ_SUCCESS)

return -3;

}

/* Add UDP transport. */

{

pjsua_transport_config cfg;

pjsua_transport_config_default(&cfg);

cfg.port = 5060;

status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);

if (status != PJ_SUCCESS)

return -4;

}

/* Initialization is done, now start pjsua */

status = pjsua_start();

if (status != PJ_SUCCESS)

return -5;

}

void qt_sip::on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,pjsip_rx_data *rdata)

{

pjsua_call_info ci;

PJ_UNUSED_ARG(acc_id);

PJ_UNUSED_ARG(rdata);

pjsua_call_get_info(call_id, &ci);

PJ_LOG(3,(THIS_FILE, "Incoming call from %.*s!!",

(int)ci.remote_info.slen,

ci.remote_info.ptr));

/* Automatically answer incoming calls with 200/OK */

pjsua_call_answer(call_id, 200, NULL, NULL);

}

/* Callback called by the library when call's state has changed */

void qt_sip::on_call_state(pjsua_call_id call_id, pjsip_event *e)

{

pjsua_call_info ci;

PJ_UNUSED_ARG(e);

pjsua_call_get_info(call_id, &ci);

PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id,

(int)ci.state_text.slen,

ci.state_text.ptr));

}

/* Callback called by the library when call's media state has changed */

void qt_sip::on_call_media_state(pjsua_call_id call_id)

{

pjsua_call_info ci;

pjsua_call_get_info(call_id, &ci);

if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {

// When media is active, connect call to sound device.

pjsua_conf_connect(ci.conf_slot, 0);

pjsua_conf_connect(0, ci.conf_slot);

}

/* Display error and exit application */

void qt_sip::error_exit(const char *title, pj_status_t status)

{

pjsua_perror(THIS_FILE, title, status);

pjsua_destroy();

exit(1);

}

注册

说明：

在封装sip协议栈的操作接口时，最好是使用SIP协议栈提供的接口，比如：

使用pj_ansi_snprintf这个函数，而不直接使用C的snprintf或是sprintf函数。

int qt_sip::sip_register(char *cname,char *cpasswd,char *sdomain)

{

pj_status_t status;

/* Register to SIP server by creating SIP account. */

{

pjsua_acc_config cfg;

pjsua_acc_config_default(&cfg);

char uri[PJSIP_MAX_URL_SIZE];

//local地址

//minisipserver的ip地址：100.100.100.199

//在minisipserver服务器上添加SIP账号，名称：100，密码：100

//形式："sip:账号名称@服务器地址"

pj_ansi_snprintf(uri, PJSIP_MAX_URL_SIZE,

"sip:%s@%s",

cname,sdomain);

cfg.id = pj_str(uri);

//服务器地址

/ /形式：“sip:地址”

//“sip:100.100.100.199”

char reg_uri[PJSIP_MAX_URL_SIZE];

pj_ansi_snprintf(reg_uri, PJSIP_MAX_URL_SIZE,"sip:%s",sdomain);

PJ_LOG(3,(THIS_FILE, "***%s***",reg_uri));

cfg.reg_uri = pj_str(reg_uri);

cfg.cred_count = 1;

char realm_uri[PJSIP_MAX_URL_SIZE];

pj_ansi_snprintf(realm_uri,100,"%s",sdomain);

PJ_LOG(3,(THIS_FILE, "***%s***",realm_uri));

cfg.cred_info[0].realm = pj_str("*");

//cfg.cred_info[0].scheme = pj_str("digest");

cfg.cred_info[0].username = pj_str(cname);

cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;

cfg.cred_info[0].data = pj_str(cpasswd);

status = pjsua_acc_add(&cfg, PJ_TRUE, &g_acc_id);

if (status != PJ_SUCCESS)

return -6;

}

return 0;

}

拨号

//在minisipserver服务器上添加SIP账号，名称：101

//使用另外的客户端注册

//拨打101，命令格式：“sip:101@100.100.100.199”

int qt_sip::sip_dial(char *phonenum)

{

pj_status_t status;

char tmp_uri[PJSIP_MAX_URL_SIZE];

pj_ansi_snprintf(tmp_uri, PJSIP_MAX_URL_SIZE,

"sip:%s@%s",

phonenum,g_server_name);

/* If argument is specified, it's got to be a valid SIP URL */

status = pjsua_verify_url(tmp_uri);

if (status != PJ_SUCCESS)

return -1;

pj_str_t uri = pj_str(tmp_uri);

status = pjsua_call_make_call(g_acc_id, &uri, 0, NULL, NULL, NULL);

if (status != PJ_SUCCESS)

return -2;

return 0;

}

挂断

void qt_sip::sip_hangup()

{

pjsua_call_hangup_all();

}

打印日志：

1）初始化

[root@FriendlyARM QT_PJSIP_VOIP-build-desktop-Qt_4_7_0__QtEmbedded-4_7_0-arm____]# ./QTVOIP -qws

11:56:48.281 os_core_unix.c !pjlib 2.1 for POSIX initialized

11:56:58.294 sip_endpoint.c .Creating endpoint instance...

11:56:58.295 pjlib .select() I/O Queue created (0x1b16d8)

11:56:58.295 sip_endpoint.c .Module "mod-msg-print" registered

11:56:58.295 sip_transport. .Transport manager created.

11:56:58.295 pjsua_core.c .PJSUA state changed: NULL --> CREATED

11:56:58.295 sip_endpoint.c .Module "mod-pjsua-log" registered

11:56:58.295 sip_endpoint.c .Module "mod-tsx-layer" registered

11:56:58.295 sip_endpoint.c .Module "mod-stateful-util" registered

11:56:58.295 sip_endpoint.c .Module "mod-ua" registered

11:56:58.295 sip_endpoint.c .Module "mod-100rel" registered

11:56:58.295 sip_endpoint.c .Module "mod-pjsua" registered

11:56:58.296 sip_endpoint.c .Mo[19531.501318] EPLL Rate changes from 49152000 to 67738000

[19531.503760] EPLL Rate changes from 67738000 to 49152000

dule "mod-invite" registered

11:56:58.364 pa_dev.c ..PortAudio sound library initialized, status=0

11:56:58.364 pa_dev.c ..PortAudio host api count=2

11:56:58.364 pa_dev.c ..Sound device count=4

11:56:58.364 pjlib ..select() I/O Queue created (0x1c947c)

11:56:58.396 sip_endpoint.c .Module "mod-evsub" registered

11:56:58.396 sip_endpoint.c .Module "mod-presence" registered

11:56:58.397 sip_endpoint.c .Module "mod-mwi" registered

11:56:58.397 sip_endpoint.c .Module "mod-refer" registered

11:56:58.397 sip_endpoint.c .Module "mod-pjsua-pres" registered

11:56:58.397 sip_endpoint.c .Module "mod-pjsua-im" registered

11:56:58.397 sip_endpoint.c .Module "mod-pjsua-options" registered

11:56:58.397 pjsua_core.c .1 SIP worker threads created

11:56:58.397 pjsua_core.c .pjsua version 2.1 for Linux-3.0.8/armv7l/glibc-2.11 initialized

11:56:58.397 pjsua_core.c .PJSUA state changed: CREATED --> INIT

11:57:08.402 pjsua_core.c SIP UDP socket reachable at 100.100.100.102:5060

11:57:08.402 udp0x1d7e78 SIP UDP transport started, published address is 100.100.100.102:5060

11:57:08.402 pjsua_core.c PJSUA state changed: INIT --> STARTING

11:57:08.402 sip_endpoint.c .Module "mod-unsolicited-mwi" registered

11:57:08.402 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING

2）注册

11:59:02.256 APP ***sip:100.100.100.199***

11:59:02.256 APP ***100.100.100.199***

11:59:02.256 pjsua_acc.c Adding account: id=sip:100@100.100.100.199

11:59:02.256 pjsua_acc.c .Account sip:100@100.100.100.199 added with id 0

11:59:02.256 pjsua_acc.c .Acc 0: setting registration..

11:59:02.257 pjsua_core.c ...TX 506 bytes Request msg REGISTER/cseq=44772 (tdta0x2b89e8) to UDP 100.100.100.199:5060:

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPj82a14091-f95e-44de-b525-fe21626f594b

Max-Forwards: 70

From: <sip:100@100.100.100.199>;tag=b6314bb4-660d-487b-b2f0-8e7099dbd28a

To: <sip:100@100.100.100.199>

Call-ID: c90be808-43c1-498a-9817-1a4c53b06c4a

CSeq: 44772 REGISTER

Contact: <sip:100@100.100.100.102:5060;ob>

Expires: 300

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Content-Length: 0

--end msg--

11:59:02.258 pjsua_acc.c ..Acc 0: Registration sent

11:59:02.271 pjsua_core.c .RX 595 bytes Response msg 407/REGISTER/cseq=44772 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPj82a14091-f95e-44de-b525-fe21626f594b

From: <sip:100@100.100.100.199>;tag=b6314bb4-660d-487b-b2f0-8e7099dbd28a

To: <sip:100@100.100.100.199>;tag=2db82286

CSeq: 44772 REGISTER

Call-ID: c90be808-43c1-498a-9817-1a4c53b06c4a

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Proxy-Authenticate: Digest realm="example.com",algorithm=MD5,nonce="649b3b454ccd12ff21c6509715c2513b",stale=false

Content-Length: 0

--end msg--

11:59:02.272 pjsua_core.c ....TX 700 bytes Request msg REGISTER/cseq=44773 (tdta0x2b89e8) to UDP 100.100.100.199:5060:

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPj2ac23c60-15b7-475f-abd2-247976b94d3e

Max-Forwards: 70

From: <sip:100@100.100.100.199>;tag=b6314bb4-660d-487b-b2f0-8e7099dbd28a

To: <sip:100@100.100.100.199>

Call-ID: c90be808-43c1-498a-9817-1a4c53b06c4a

CSeq: 44773 REGISTER

Contact: <sip:100@100.100.100.102:5060;ob>

Expires: 300

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Proxy-Authorization: Digest username="100", realm="example.com", nonce="649b3b454ccd12ff21c6509715c2513b", uri="sip:100.100.100.199", response="22b3e8e9d88a2f9ace62c83308c7d741", algorithm=MD5

Content-Length: 0

--end msg--

11:59:02.273 pjsua_core.c .RX 503 bytes Response msg 200/REGISTER/cseq=44773 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPj2ac23c60-15b7-475f-abd2-247976b94d3e

From: <sip:100@100.100.100.199>;tag=b6314bb4-660d-487b-b2f0-8e7099dbd28a

To: <sip:100@100.100.100.199>;tag=2db82286

CSeq: 44773 REGISTER

Call-ID: c90be808-43c1-498a-9817-1a4c53b06c4a

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Contact: <sip:100@100.100.100.102>

Expires: 300

Content-Length: 0

--end msg--

11:59:02.273 pjsua_acc.c ....SIP outbound status for acc 0 is not active

11:59:02.273 pjsua_acc.c ....sip:100@100.100.100.199: registration success, status=200 (OK), will re-register in 300 seconds

11:59:02.273 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:100.100.100.199:5060, interval:15s

3）拨号

12:00:10.864 pjsua_call.c !Making call with acc #0 to sip:101@100.100.100.199

12:00:10.864 pjsua_aud.c .Set sound device: capture=-1, playback=-2

12:00:10.865 pjsua_aud.c ..Opening sound device PCM@16000/1/20ms

Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294

Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1873

Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994

12:00:10.876 ec0x2beb68 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms

12:00:10.877 pjsua_media.c .Call 0: initializing media..

12:00:15.893 sound_port.c EC suspended because of inactivity

12:00:20.886 pjsua_media.c !..RTP socket reachable at 100.100.100.102:40000

12:00:20.887 pjsua_media.c ..RTCP socket reachable at 100.100.100.102:40001

12:00:20.887 pjsua_media.c ..Media index 0 selected for audio call 0

12:00:20.888 pjsua_core.c ....TX 1088 bytes Request msg INVITE/cseq=9089 (tdta0x2f2898) to UDP 100.100.100.199:5060:

INVITE sip:101@100.100.100.199 SIP/2.0

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

Max-Forwards: 70

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199

Contact: <sip:100@100.100.100.102:5060;ob>

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

CSeq: 9089 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Session-Expires: 1800

Min-SE: 90

Content-Type: application/sdp

Content-Length: 484

v=0

o=- 3541291220 3541291220 IN IP4 100.100.100.102

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96

c=IN IP4 100.100.100.102

b=TIAS:64000

a=rtcp:40001 IN IP4 100.100.100.102

a=sendrecv

a=rtpmap:98 speex/16000

a=rtpmap:97 speex/8000

a=rtpmap:99 speex/32000

a=rtpmap:104 iLBC/8000

a=fmtp:104 mode=30

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

--end msg--

12:00:20.888 APP .......Call 0 state=CALLING

12:00:20.897 pjsua_core.c .RX 486 bytes Response msg 100/INVITE/cseq=9089 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199;tag=2a46556f

CSeq: 9089 INVITE

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Contact: <sip:101@100.100.100.199>

Content-Length: 0

--end msg--

12:00:20.907 pjsua_core.c .RX 487 bytes Response msg 180/INVITE/cseq=9089 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199;tag=2a46556f

CSeq: 9089 INVITE

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Contact: <sip:101@100.100.100.199>

Content-Length: 0

--end msg--

12:00:20.907 APP .....Call 0 state=EARLY

4）挂断

12:01:03.160 pjsua_call.c !Hangup all calls..

12:01:03.160 pjsua_call.c .Call 0 hanging up: code=0..

12:01:03.160 pjsua_core.c .....TX 345 bytes Request msg CANCEL/cseq=9089 (tdta0x2a7e00) to UDP 100.100.100.199:5060:

CANCEL sip:101@100.100.100.199 SIP/2.0

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

Max-Forwards: 70

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

CSeq: 9089 CANCEL

Content-Length: 0

--end msg--

12:01:03.162 pjsua_core.c .RX 482 bytes Response msg 200/CANCEL/cseq=9089 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199;tag=2a46556f

CSeq: 9089 CANCEL

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Contact: <sip:101@100.100.100.199>

Content-Length: 0

--end msg--

12:01:03.162 pjsua_core.c .RX 498 bytes Response msg 487/INVITE/cseq=9089 (rdata0x1d92a4) from UDP 100.100.100.199:5060:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199;tag=2a46556f

CSeq: 9089 INVITE

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

Allow: REGISTER,INVITE,OPTIONS,ACK,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE

User-Agent: miniSipServer V2.3.2 (100) build on Nov 6 2008

Contact: <sip:101@100.100.100.199>

Content-Length: 0

--end msg--

12:01:03.162 pjsua_core.c ..TX 352 bytes Request msg ACK/cseq=9089 (tdta0x2f71a0) to UDP 100.100.100.199:5060:

ACK sip:101@100.100.100.199 SIP/2.0

Via: SIP/2.0/UDP 100.100.100.102:5060;rport;branch=z9hG4bKPjb02d3774-83fd-4ff5-8694-26d12697d8d3

Max-Forwards: 70

From: sip:100@100.100.100.199;tag=eb1a9606-dde3-42d8-9253-955509e34509

To: sip:101@100.100.100.199;tag=2a46556f

Call-ID: b30d7a62-82ce-4a1c-8927-fd69d15c8b3b

CSeq: 9089 ACK

Content-Length: 0

--end msg--

12:01:03.163 APP .....Call 0 state=DISCONNCTD

12:01:03.163 pjsua_media.c .....Call 0: deinitializing media..

12:01:04.163 pjsua_aud.c Closing sound device after idle for 1 second(s)

12:01:04.163 pjsua_aud.c .Closing mini210: (hw:0,0) sound playback device and mini210: (hw:0,0) sound capture device

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