1,Linphone初始化工作;
入口:linphone_core_new(...) -->linphone_core_init(core,vtable,config_path, factory_config_path, userdata);
I,首先就是与oRTP(基于RFC3550的一个实现)协议栈相关的初始化操作:如:ortp_init();
在这个函数里面做的工作有:
[A]av_profile_init()即负载类型的初始化。rtp最大支持128种负载类型。这里有个概念:
RtpProfile:
* The RTP profile is a table RTP_PROFILE_MAX_PAYLOADS entries to make the matching
* between RTP payload type number and the PayloadType that defines the type of
* media.
[B]rtp全局统计信息初始化
typedef struct rtp_stats
{
uint64_t packet_sent;
uint64_t sent; /* bytes sent */
uint64_t recv; /* bytes of payload received and delivered in time to the application */
uint64_t hw_recv; /* bytes of payload received */
uint64_t packet_recv; /* number of packets received */
uint64_t outoftime; /* number of packets that were received too late */
uint64_t cum_packet_loss; /* cumulative number of packet lost */
uint64_t bad; /* packets that did not appear to be RTP */
uint64_t discarded; /* incoming packets discarded because the queue exceeds its max size */
uint64_t sent_rtcp_packets; /* sent RTCP packets counter (only packets that embed a report block are considered) */
} rtp_stats_t;
接下来是Linphone所用到的一些负载类型的初始话(assign),包括音频,视频的,与number相关的及无关的两类;
II, 其次是mediastream2的一些初始化。ms_init();
包括日志相关的设置,ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
ortp_set_log_handler(ms_android_log_handler);
Filter初始化/注册:
/* register builtin MSFilter's */
for (i=0;ms_filter_descs[i]!=NULL;i++){
ms_filter_register(ms_filter_descs[i]);
}
声卡初始化,
cm=ms_snd_card_manager_get();
for (i=0;ms_snd_card_descs[i]!=NULL;i++){
ms_snd_card_manager_register_desc(cm,ms_snd_card_descs[i]);
}
网络摄像头的初始化,
MSWebCamManager *wm;
wm=ms_web_cam_manager_get();
for (i=0;ms_web_cam_descs[i]!=NULL;i++){
ms_web_cam_manager_register_desc(wm,ms_web_cam_descs[i]);
}
绘制视频图像初始化(opengl):
libmsandroidopengldisplay_init();
还要初始化一个mediastream2的事件队列:ms_event_queue_new();
III,再次就是初始化一个很重要的结构体对象:
struct Sal{
SalCallbacks callbacks;
MSList *calls; /*MSList of SalOp */
MSList *registers;/*MSList of SalOp */
MSList *out_subscribes;/*MSList of SalOp */
MSList *in_subscribes;/*MSList of SalOp */
MSList *pending_auths;/*MSList of SalOp */
MSList *other_transactions; /*MSList of SalOp */
int running;
int session_expires;
int keepalive_period;
void *up;
bool_t one_matching_codec;
bool_t double_reg;
bool_t use_rports;
bool_t use_101;
bool_t reuse_authorization;
char* rootCa; /* File _or_ folder containing root CA */
};
这个对象很重要,在全局只有一个。是sip信号处理抽象层;初始化它的同时进行exosip的初始化:eXosip_init();
并设置协议信号处理回调函数结构体对象:linphone_sal_callbacks,用以处理各种sip消息。
IV,最后就是初始化一些配置信息:
sip_setup_register_all(); //linphone_sip_login 【注册上去】。
sound_config_read(lc);
net_config_read(lc);
rtp_config_read(lc);
codecs_config_read(lc);
sip_config_read(lc); /* this will start eXosip*/
IP,端口,协议配置。
linphone_core_set_sip_transports(lc,&tr);监听
sal_root_ca();认证
代理配置信息:lc->sip_conf.proxies
默认代理:linphone_core_set_default_proxy_index();
授权信息:
/* read authentication information */
for(i=0;; i++){
LinphoneAuthInfo *ai=linphone_auth_info_new_from_config_file(lc->config,i);
if (ai!=NULL){
linphone_core_add_auth_info(lc,ai);
linphone_auth_info_destroy(ai);
}else{
break;
}
}
其它配置信息:/*for tuning or test*/等等。。。
}
video_config_read(lc);
Linphone 进入“Read” 状态。。。。。。
2,注册到服务器的过程。
在linphone_core_iterate()中 proxy_update(lc); 就是注册的触发函数,依次循环每个代理配置,判断commit=true && reg_sendregistry = true;
I,linphone_proxy_config_activate_sip_setup:激活一个sipsetup环境
II,linphone_proxy_config_register(LinphoneProxyConfig *cfg);生成一个用于注册的SalOp,并设置其contact和
user_pointer 如:
sal_op_set_contact(obj->op,contact);
ms_free(contact);
sal_op_set_user_pointer(obj->op,obj);
然后发出注册消息:sal_register(obj->op,obj->reg_proxy,obj->reg_identity,obj->expires),开始了注册过程...
首次注册没有授权信息,所以会失败,收到sip消息: EXOSIP_REGISTRATION_FAILURE:
case 401:case 407:process_authentication(sal,ev);就会提出授权要求(有sip信号处理回调函数来处理):auth_requested,并添加到:op->pending_auth=ev;
如果其它原因的错误就需要其它的处理了。比如case 423:case 606,当用户收到授权请求时,就会判断当前的授权信息是否满足,
LinphoneAuthInfo *ai=(LinphoneAuthInfo*)linphone_core_find_auth_info(lc,realm,username);然后给注册操作授权:sal_op_authenticate(h,&sai);
如果当前没有满足的授权信息,则可能需要用户输入授权信息。。。。
如果注册成功:则需要首先提示授权工作了authentication_ok(sal,ev)。再确定回应里面的请求-是否需要重新注册新的contact.(register_again_with_updated_contact),
如果需要,就重新注册,update_contact_from_response(op,last_answer);contact do not match, need to update the register ??。。。。
不需要的话,就可以提示注册成功的消息了。至此,注册完毕;
3,一次呼叫建立的过程(重点,内容很多呀...);
LinphoneCall * linphone_core_invite(LinphoneCore *lc, const char *url) :
LinphoneCall * linphone_core_invite(LinphoneCore *lc, const char *url);
LinphoneCall * linphone_core_invite_address(LinphoneCore *lc, const LinphoneAddress *addr);
LinphoneCall * linphone_core_invite_with_params(LinphoneCore *lc, const char *url, const LinphoneCallParams *params);
LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const LinphoneAddress *addr, const LinphoneCallParams *params);
四个出发函数。最终要依赖linphone_core_invite_address_with_params,中间可能多一些地址转换,呼叫参数(如是否支持视频等)的初始化工作。
进入到呼叫函数linphone_core_invite_address_with_params后,首先判断当前是否有呼叫,以及是否达到呼叫数目的最大限。
linphone_core_in_call(lc);
linphone_core_can_we_add_call(lc);
接这对默认代理,和呼叫地址中的代理进行匹配,如果不一样,则进行重写默认的代理,以呼叫地址中的代理为准。生成一个from字符串。
如果都为空则from=linphone_core_get_primary_contact(lc);/* if no proxy or no identity defined for this proxy, default to primary contact*/
生成URL:parsed_url2=linphone_address_new(from);创建一个新的Call:
call=linphone_call_new_outgoing(lc,parsed_url2,linphone_address_clone(addr),params);里面包括一些设置:
linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
linphone_call_init_common(call,from,to);//在这里计数:refcnt=1,设置引用基数.
call->params=*params;
call->localdesc=create_local_media_description (lc,call); //本地媒体类型描述
call->camera_active=params->has_video;
if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
linphone_core_run_stun_tests(call->core,call); //防火墙策略
discover_mtu(lc,linphone_address_get_domain (to));
if (params->referer){
sal_call_set_referer (call->op,params->referer->op); //是呼叫转移吗
}
设置route:sal_op_set_route(call->op,route);添加到LinphoneCore:linphone_core_add_call(lc,call),然后lc->current_call=call;
接下来,如果需要ping
就可以直接进行呼叫了:linphone_core_start_invite(lc,call,dest_proxy);不然就接着进行ping操作:
if (dest_proxy!=NULL || lc->sip_conf.ping_with_options==FALSE){
linphone_core_start_invite(lc,call,dest_proxy);
}else{
/*defer the start of the call after the OPTIONS ping*/
call->ping_op=sal_op_new(lc->sal);
sal_ping(call->ping_op,from,real_url); ///ping操作,,,eXosip_options_build_request -->...--> eXosip_options_send_request(options);
sal_op_set_user_pointer(call->ping_op,call);
call->start_time=time(NULL);
}
linphone_core_iterator里面如果curtime-call->start_time>=2,则会不等ping回来,就呼叫linphone_core_start_invite(lc,call,NULL);
如果需要ping操作,就需要处理Ping_op的回应,消息处理函数是:other_request_reply(sal,ev);
-->sal->callbacks.ping_reply(op);在ping_reply里面:linphone_core_start_invite(call->core,call,NULL);
至此,终于可以进行呼叫了。当对初始化了的call进行,真正呼叫时要什么呢?
I,为呼叫操作设置contact,sal_op_set_contact(call->op, contact);而这里的contact是从get_fixed_contact(lc,call,dest_proxy),即dest_proxy而来的。
II,linphone_call_init_media_streams(call);初始化音频,视频的媒体流,
第一部分:初始化 audio_stream_new ,统计信息初始化:ms_filter_enable_statistics(TRUE);ms_filter_reset_statistics();
创建配置session:stream->session=create_duplex_rtpsession(locport,ipv6);//RTP_SESSION_SENDRECV 模式的
rtpr=rtp_session_new(RTP_SESSION_SENDRECV);[[[===
rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
rtp_session_set_scheduling_mode(rtpr,0);
rtp_session_set_blocking_mode(rtpr,0);
rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
rtp_session_set_symmetric_rtp(rtpr,TRUE);
rtp_session_set_local_addr(rtpr,ipv6 ? "::" : "0.0.0.0",locport);
rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
rtp_session_set_ssrc_changed_threshold(rtpr,0);
rtp_session_set_rtcp_report_interval(rtpr,2500); /*at the beginning of the session send more reports*/
disable_checksums(rtp_session_get_rtp_socket(rtpr));
添加发送的filter:stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
添加回声消除的filter:stream->ec=ms_filter_new_from_desc(ec_desc);
生成并初始化,为本stream注册rtp事件队列:stream->evq=ortp_ev_queue_new();rtp_session_register_event_queue(stream->session,stream->evq);
其它初始化:stream->play_dtmfs=TRUE;
stream->use_gc=FALSE;
stream->use_agc=FALSE;
stream->use_ng=FALSE;
===]]]
接着,如果支持回声限制,则根据配置信息设置一些相关参数,如:audio_stream_enable_echo_limiter(audiostream,ELControlFull);
接着,如果支持回声消除,则根据配置信息设置一些相关参数,如:audio_stream_set_echo_canceller_params;
接着,是否支持获取控制,以便不获取噪声,有个噪声的gateway要设置:
int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
audio_stream_enable_noise_gate(audiostream,enabled);
在就是为session设置if (lc->a_rtp) rtp_session_set_transports(audiostream->session,lc->a_rtp,lc->a_rtcp);
给Call也注册一个ort事件队列:call->audiostream_app_evq = ortp_ev_queue_new();
rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
第二部分:如果支持视频,则需要call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));//初始化视频流
具体:VideoStream *stream = (VideoStream *)ms_new0 (VideoStream, 1);
stream->session=create_duplex_rtpsession(locport,use_ipv6);
stream->evq=ortp_ev_queue_new();
stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
rtp_session_register_event_queue(stream->session,stream->evq);
stream->sent_vsize.width=MS_VIDEO_SIZE_CIF_W;
stream->sent_vsize.height=MS_VIDEO_SIZE_CIF_H;
stream->dir=VideoStreamSendRecv;
choose_display_name(stream);
接下来:设置display_filter_name,设置video_stream_set_event_callback,设置rtp_session_set_transports,
注册rtp事件队列:rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
III,sal_call_set_local_media_description ,设置本地媒体格式描述。
解析地址from ,url,等,然后开始真正呼叫。。err=sal_call(call->op,from,real_url);具体展开。。。:
int sal_call(SalOp *h, const char *from, const char *to){
int err;
osip_message_t *invite=NULL;
sal_op_set_from(h,from);
sal_op_set_to(h,to);
sal_exosip_fix_route(h);
err=eXosip_call_build_initial_invite(&invite,to,from,sal_op_get_route(h),"Phone call");
if (err!=0){
ms_error("Could not create call.");
return -1;
}
osip_message_set_allow(invite, "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO");
if (h->base.contact){
_osip_list_set_empty(&invite->contacts,(void (*)(void*))osip_contact_free);
osip_message_set_contact(invite,h->base.contact);
}
if (h->base.root->session_expires!=0){
osip_message_set_header(invite, "Session-expires", "200");
osip_message_set_supported(invite, "timer");
}
if (h->base.local_media){
h->sdp_offering=TRUE;
set_sdp_from_desc(invite,h->base.local_media);
}else h->sdp_offering=FALSE;
if (h->replaces){
osip_message_set_header(invite,"Replaces",h->replaces);
if (h->referred_by)
osip_message_set_header(invite,"Referred-By",h->referred_by);
}
eXosip_lock();
err=eXosip_call_send_initial_invite(invite);
eXosip_unlock();
h->cid=err;
if (err<0){
ms_error("Fail to send invite !");
return -1;
}else{
sal_add_call(h->base.root,h);//把操作添加到sal中....
}
return 0;
}
最后设置状态:Contacting。。。
barmsg=ortp_strdup_printf("%s %s", _("Contacting"), real_url);
if (lc->vtable.display_status!=NULL)
lc->vtable.display_status(lc,barmsg);
如此,便开始了等待代理服务器返回消息的状态了。。。。
接下来分析,当接受到对放来的call-request的时候,怎么处理。。。。
case EXOSIP_CALL_INVITE:表示收到了一个呼叫的消息。执行:inc_new_call(Sal *sal, eXosip_event_t *ev);
首先为这个消息生成一个SalOp操作。得到sdp信息:eXosip_get_sdp_info(ev->request);接着从request里面获取一些参数如:origin
ua,replaces,from ,to ,sdp,call_info,tid,cid, did,等等,然后进入sip消息回调里面的call_received回调进行处理,进入这个函数后就是
和主动呼叫(outgoingcall)的模式很相像了,判断收处于呼叫状态,是否达到最大呼叫数目linphone_core_can_we_add_call,有传入的salop得到from
和to,进而判断是否是重复呼叫is_duplicate_call?如果都满足上述条件,就可以惊醒呼叫的创建了:call=linphone_call_new_incoming(lc,from_addr,to_addr,h);
里面包括判决是否发ping指令,本地medaidesc,linphone_call_init_common,create_local_media_description,以及
linphone_core_get_firewall_policy等等。。。,随后开始进行sdp,媒体类型协商:sal_call_get_final_media_description-->sdp_process,
最后添加call,linphone_core_add_call,提示电话来了的消息: lc->vtable.display_status(lc,barmesg); //XX is contacting you?
与此同时,开始ring_start(),播放ringback(/* play the ring if this is the only call*/),
同时发180消息sal_call_notify_ringing(h,propose_early_media || ringback_tone!=NULL);如果自动应答模式,还需接受call.linphone_core_accept_call(lc,call);
具体如下:
【【【---
sal_call_notify_ringing(h,propose_early_media || ringback_tone!=NULL);//180消息
if (propose_early_media || ringback_tone!=NULL){
linphone_call_set_state(call,LinphoneCallIncomingEarlyMedia,"Incoming call early media");
linphone_core_update_streams(lc,call,md);
}
if (sal_call_get_replaces(call->op)!=NULL && lp_config_get_int(lc->config,"sip","auto_answer_replacing_calls",1)){
linphone_core_accept_call(lc,call);
}
---】】】
在换回来说吧,当主叫一方收到180Ring的消息后(case EXOSIP_CALL_RINGING),进入到函数:call_ringing(Sal *sal, eXosip_event_t *ev)
首先:call_proceeding(Sal *sal, eXosip_event_t *ev) --->
/* update contact if received and rport are set by the server
note: will only be used by remote for next INVITE, if any...*/
update_contact_from_response(op,ev->response);
然后得到sdp:sdp=eXosip_get_sdp_info(ev->response);生成本地的mediadesc,然后协商出一个result md,if (op->base.local_media) sdp_process(op);
回调到sip消息回调处理函数里面sal->callbacks.call_ringing(op);,通知界面,开始early medai,ringing了。。。linphone_core_update_streams,即update一下。
又回去,当被叫决定答应呼叫是,他会调用int linphone_core_accept_call(LinphoneCore *lc, LinphoneCall *call);
里面会做:I,/* check if this call is supposed to replace an already running one*/replaced=sal_call_get_replaces(call->op);
II,/*try to be best-effort in giving real local or routable contact address*/ --->sal_op_set_contact(call->op,contact);
III,/*stop ringing */-->ring_stop(lc->ringstream);
IV,if (call->audiostream==NULL)
linphone_call_init_media_streams(call); //这个之前已经做过了,这里只是检查做没做,谨慎期间。
IIV,sal_call_accept(call->op);//发送200OK消息
IIIV,new_md=sal_call_get_final_media_description(call->op);
linphone_core_update_streams(lc, call, new_md); //更新媒体流。。。
VI,通知:ms_message("call answered.");
4,
到这里,一个呼叫的整个常规流程也就完事儿了。而整个linphone的主要功能框架也基本完成。但中间一些细节,如如何添加每个节点的filter,如何维持更新媒体流,借助与mediastream2,怎
么实现流媒体的过程,还需进一步分析。(2011-11-10 宣继托)
入口:linphone_core_new(...) -->linphone_core_init(core,vtable,config_path, factory_config_path, userdata);
I,首先就是与oRTP(基于RFC3550的一个实现)协议栈相关的初始化操作:如:ortp_init();
在这个函数里面做的工作有:
[A]av_profile_init()即负载类型的初始化。rtp最大支持128种负载类型。这里有个概念:
RtpProfile:
* The RTP profile is a table RTP_PROFILE_MAX_PAYLOADS entries to make the matching
* between RTP payload type number and the PayloadType that defines the type of
* media.
[B]rtp全局统计信息初始化
typedef struct rtp_stats
{
uint64_t packet_sent;
uint64_t sent; /* bytes sent */
uint64_t recv; /* bytes of payload received and delivered in time to the application */
uint64_t hw_recv; /* bytes of payload received */
uint64_t packet_recv; /* number of packets received */
uint64_t outoftime; /* number of packets that were received too late */
uint64_t cum_packet_loss; /* cumulative number of packet lost */
uint64_t bad; /* packets that did not appear to be RTP */
uint64_t discarded; /* incoming packets discarded because the queue exceeds its max size */
uint64_t sent_rtcp_packets; /* sent RTCP packets counter (only packets that embed a report block are considered) */
} rtp_stats_t;
接下来是Linphone所用到的一些负载类型的初始话(assign),包括音频,视频的,与number相关的及无关的两类;
II, 其次是mediastream2的一些初始化。ms_init();
包括日志相关的设置,ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
ortp_set_log_handler(ms_android_log_handler);
Filter初始化/注册:
/* register builtin MSFilter's */
for (i=0;ms_filter_descs[i]!=NULL;i++){
ms_filter_register(ms_filter_descs[i]);
}
声卡初始化,
cm=ms_snd_card_manager_get();
for (i=0;ms_snd_card_descs[i]!=NULL;i++){
ms_snd_card_manager_register_desc(cm,ms_snd_card_descs[i]);
}
网络摄像头的初始化,
MSWebCamManager *wm;
wm=ms_web_cam_manager_get();
for (i=0;ms_web_cam_descs[i]!=NULL;i++){
ms_web_cam_manager_register_desc(wm,ms_web_cam_descs[i]);
}
绘制视频图像初始化(opengl):
libmsandroidopengldisplay_init();
还要初始化一个mediastream2的事件队列:ms_event_queue_new();
III,再次就是初始化一个很重要的结构体对象:
struct Sal{
SalCallbacks callbacks;
MSList *calls; /*MSList of SalOp */
MSList *registers;/*MSList of SalOp */
MSList *out_subscribes;/*MSList of SalOp */
MSList *in_subscribes;/*MSList of SalOp */
MSList *pending_auths;/*MSList of SalOp */
MSList *other_transactions; /*MSList of SalOp */
int running;
int session_expires;
int keepalive_period;
void *up;
bool_t one_matching_codec;
bool_t double_reg;
bool_t use_rports;
bool_t use_101;
bool_t reuse_authorization;
char* rootCa; /* File _or_ folder containing root CA */
};
这个对象很重要,在全局只有一个。是sip信号处理抽象层;初始化它的同时进行exosip的初始化:eXosip_init();
并设置协议信号处理回调函数结构体对象:linphone_sal_callbacks,用以处理各种sip消息。
IV,最后就是初始化一些配置信息:
sip_setup_register_all(); //linphone_sip_login 【注册上去】。
sound_config_read(lc);
net_config_read(lc);
rtp_config_read(lc);
codecs_config_read(lc);
sip_config_read(lc); /* this will start eXosip*/
IP,端口,协议配置。
linphone_core_set_sip_transports(lc,&tr);监听
sal_root_ca();认证
代理配置信息:lc->sip_conf.proxies
默认代理:linphone_core_set_default_proxy_index();
授权信息:
/* read authentication information */
for(i=0;; i++){
LinphoneAuthInfo *ai=linphone_auth_info_new_from_config_file(lc->config,i);
if (ai!=NULL){
linphone_core_add_auth_info(lc,ai);
linphone_auth_info_destroy(ai);
}else{
break;
}
}
其它配置信息:/*for tuning or test*/等等。。。
}
video_config_read(lc);
Linphone 进入“Read” 状态。。。。。。
2,注册到服务器的过程。
在linphone_core_iterate()中 proxy_update(lc); 就是注册的触发函数,依次循环每个代理配置,判断commit=true && reg_sendregistry = true;
I,linphone_proxy_config_activate_sip_setup:激活一个sipsetup环境
II,linphone_proxy_config_register(LinphoneProxyConfig *cfg);生成一个用于注册的SalOp,并设置其contact和
user_pointer 如:
sal_op_set_contact(obj->op,contact);
ms_free(contact);
sal_op_set_user_pointer(obj->op,obj);
然后发出注册消息:sal_register(obj->op,obj->reg_proxy,obj->reg_identity,obj->expires),开始了注册过程...
首次注册没有授权信息,所以会失败,收到sip消息: EXOSIP_REGISTRATION_FAILURE:
case 401:case 407:process_authentication(sal,ev);就会提出授权要求(有sip信号处理回调函数来处理):auth_requested,并添加到:op->pending_auth=ev;
如果其它原因的错误就需要其它的处理了。比如case 423:case 606,当用户收到授权请求时,就会判断当前的授权信息是否满足,
LinphoneAuthInfo *ai=(LinphoneAuthInfo*)linphone_core_find_auth_info(lc,realm,username);然后给注册操作授权:sal_op_authenticate(h,&sai);
如果当前没有满足的授权信息,则可能需要用户输入授权信息。。。。
如果注册成功:则需要首先提示授权工作了authentication_ok(sal,ev)。再确定回应里面的请求-是否需要重新注册新的contact.(register_again_with_updated_contact),
如果需要,就重新注册,update_contact_from_response(op,last_answer);contact do not match, need to update the register ??。。。。
不需要的话,就可以提示注册成功的消息了。至此,注册完毕;
3,一次呼叫建立的过程(重点,内容很多呀...);
LinphoneCall * linphone_core_invite(LinphoneCore *lc, const char *url) :
LinphoneCall * linphone_core_invite(LinphoneCore *lc, const char *url);
LinphoneCall * linphone_core_invite_address(LinphoneCore *lc, const LinphoneAddress *addr);
LinphoneCall * linphone_core_invite_with_params(LinphoneCore *lc, const char *url, const LinphoneCallParams *params);
LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const LinphoneAddress *addr, const LinphoneCallParams *params);
四个出发函数。最终要依赖linphone_core_invite_address_with_params,中间可能多一些地址转换,呼叫参数(如是否支持视频等)的初始化工作。
进入到呼叫函数linphone_core_invite_address_with_params后,首先判断当前是否有呼叫,以及是否达到呼叫数目的最大限。
linphone_core_in_call(lc);
linphone_core_can_we_add_call(lc);
接这对默认代理,和呼叫地址中的代理进行匹配,如果不一样,则进行重写默认的代理,以呼叫地址中的代理为准。生成一个from字符串。
如果都为空则from=linphone_core_get_primary_contact(lc);/* if no proxy or no identity defined for this proxy, default to primary contact*/
生成URL:parsed_url2=linphone_address_new(from);创建一个新的Call:
call=linphone_call_new_outgoing(lc,parsed_url2,linphone_address_clone(addr),params);里面包括一些设置:
linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
linphone_call_init_common(call,from,to);//在这里计数:refcnt=1,设置引用基数.
call->params=*params;
call->localdesc=create_local_media_description (lc,call); //本地媒体类型描述
call->camera_active=params->has_video;
if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
linphone_core_run_stun_tests(call->core,call); //防火墙策略
discover_mtu(lc,linphone_address_get_domain (to));
if (params->referer){
sal_call_set_referer (call->op,params->referer->op); //是呼叫转移吗
}
设置route:sal_op_set_route(call->op,route);添加到LinphoneCore:linphone_core_add_call(lc,call),然后lc->current_call=call;
接下来,如果需要ping
就可以直接进行呼叫了:linphone_core_start_invite(lc,call,dest_proxy);不然就接着进行ping操作:
if (dest_proxy!=NULL || lc->sip_conf.ping_with_options==FALSE){
linphone_core_start_invite(lc,call,dest_proxy);
}else{
/*defer the start of the call after the OPTIONS ping*/
call->ping_op=sal_op_new(lc->sal);
sal_ping(call->ping_op,from,real_url); ///ping操作,,,eXosip_options_build_request -->...--> eXosip_options_send_request(options);
sal_op_set_user_pointer(call->ping_op,call);
call->start_time=time(NULL);
}
linphone_core_iterator里面如果curtime-call->start_time>=2,则会不等ping回来,就呼叫linphone_core_start_invite(lc,call,NULL);
如果需要ping操作,就需要处理Ping_op的回应,消息处理函数是:other_request_reply(sal,ev);
-->sal->callbacks.ping_reply(op);在ping_reply里面:linphone_core_start_invite(call->core,call,NULL);
至此,终于可以进行呼叫了。当对初始化了的call进行,真正呼叫时要什么呢?
I,为呼叫操作设置contact,sal_op_set_contact(call->op, contact);而这里的contact是从get_fixed_contact(lc,call,dest_proxy),即dest_proxy而来的。
II,linphone_call_init_media_streams(call);初始化音频,视频的媒体流,
第一部分:初始化 audio_stream_new ,统计信息初始化:ms_filter_enable_statistics(TRUE);ms_filter_reset_statistics();
创建配置session:stream->session=create_duplex_rtpsession(locport,ipv6);//RTP_SESSION_SENDRECV 模式的
rtpr=rtp_session_new(RTP_SESSION_SENDRECV);[[[===
rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
rtp_session_set_scheduling_mode(rtpr,0);
rtp_session_set_blocking_mode(rtpr,0);
rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
rtp_session_set_symmetric_rtp(rtpr,TRUE);
rtp_session_set_local_addr(rtpr,ipv6 ? "::" : "0.0.0.0",locport);
rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
rtp_session_set_ssrc_changed_threshold(rtpr,0);
rtp_session_set_rtcp_report_interval(rtpr,2500); /*at the beginning of the session send more reports*/
disable_checksums(rtp_session_get_rtp_socket(rtpr));
添加发送的filter:stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
添加回声消除的filter:stream->ec=ms_filter_new_from_desc(ec_desc);
生成并初始化,为本stream注册rtp事件队列:stream->evq=ortp_ev_queue_new();rtp_session_register_event_queue(stream->session,stream->evq);
其它初始化:stream->play_dtmfs=TRUE;
stream->use_gc=FALSE;
stream->use_agc=FALSE;
stream->use_ng=FALSE;
===]]]
接着,如果支持回声限制,则根据配置信息设置一些相关参数,如:audio_stream_enable_echo_limiter(audiostream,ELControlFull);
接着,如果支持回声消除,则根据配置信息设置一些相关参数,如:audio_stream_set_echo_canceller_params;
接着,是否支持获取控制,以便不获取噪声,有个噪声的gateway要设置:
int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
audio_stream_enable_noise_gate(audiostream,enabled);
在就是为session设置if (lc->a_rtp) rtp_session_set_transports(audiostream->session,lc->a_rtp,lc->a_rtcp);
给Call也注册一个ort事件队列:call->audiostream_app_evq = ortp_ev_queue_new();
rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
第二部分:如果支持视频,则需要call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));//初始化视频流
具体:VideoStream *stream = (VideoStream *)ms_new0 (VideoStream, 1);
stream->session=create_duplex_rtpsession(locport,use_ipv6);
stream->evq=ortp_ev_queue_new();
stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
rtp_session_register_event_queue(stream->session,stream->evq);
stream->sent_vsize.width=MS_VIDEO_SIZE_CIF_W;
stream->sent_vsize.height=MS_VIDEO_SIZE_CIF_H;
stream->dir=VideoStreamSendRecv;
choose_display_name(stream);
接下来:设置display_filter_name,设置video_stream_set_event_callback,设置rtp_session_set_transports,
注册rtp事件队列:rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
III,sal_call_set_local_media_description ,设置本地媒体格式描述。
解析地址from ,url,等,然后开始真正呼叫。。err=sal_call(call->op,from,real_url);具体展开。。。:
int sal_call(SalOp *h, const char *from, const char *to){
int err;
osip_message_t *invite=NULL;
sal_op_set_from(h,from);
sal_op_set_to(h,to);
sal_exosip_fix_route(h);
err=eXosip_call_build_initial_invite(&invite,to,from,sal_op_get_route(h),"Phone call");
if (err!=0){
ms_error("Could not create call.");
return -1;
}
osip_message_set_allow(invite, "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO");
if (h->base.contact){
_osip_list_set_empty(&invite->contacts,(void (*)(void*))osip_contact_free);
osip_message_set_contact(invite,h->base.contact);
}
if (h->base.root->session_expires!=0){
osip_message_set_header(invite, "Session-expires", "200");
osip_message_set_supported(invite, "timer");
}
if (h->base.local_media){
h->sdp_offering=TRUE;
set_sdp_from_desc(invite,h->base.local_media);
}else h->sdp_offering=FALSE;
if (h->replaces){
osip_message_set_header(invite,"Replaces",h->replaces);
if (h->referred_by)
osip_message_set_header(invite,"Referred-By",h->referred_by);
}
eXosip_lock();
err=eXosip_call_send_initial_invite(invite);
eXosip_unlock();
h->cid=err;
if (err<0){
ms_error("Fail to send invite !");
return -1;
}else{
sal_add_call(h->base.root,h);//把操作添加到sal中....
}
return 0;
}
最后设置状态:Contacting。。。
barmsg=ortp_strdup_printf("%s %s", _("Contacting"), real_url);
if (lc->vtable.display_status!=NULL)
lc->vtable.display_status(lc,barmsg);
如此,便开始了等待代理服务器返回消息的状态了。。。。
接下来分析,当接受到对放来的call-request的时候,怎么处理。。。。
case EXOSIP_CALL_INVITE:表示收到了一个呼叫的消息。执行:inc_new_call(Sal *sal, eXosip_event_t *ev);
首先为这个消息生成一个SalOp操作。得到sdp信息:eXosip_get_sdp_info(ev->request);接着从request里面获取一些参数如:origin
ua,replaces,from ,to ,sdp,call_info,tid,cid, did,等等,然后进入sip消息回调里面的call_received回调进行处理,进入这个函数后就是
和主动呼叫(outgoingcall)的模式很相像了,判断收处于呼叫状态,是否达到最大呼叫数目linphone_core_can_we_add_call,有传入的salop得到from
和to,进而判断是否是重复呼叫is_duplicate_call?如果都满足上述条件,就可以惊醒呼叫的创建了:call=linphone_call_new_incoming(lc,from_addr,to_addr,h);
里面包括判决是否发ping指令,本地medaidesc,linphone_call_init_common,create_local_media_description,以及
linphone_core_get_firewall_policy等等。。。,随后开始进行sdp,媒体类型协商:sal_call_get_final_media_description-->sdp_process,
最后添加call,linphone_core_add_call,提示电话来了的消息: lc->vtable.display_status(lc,barmesg); //XX is contacting you?
与此同时,开始ring_start(),播放ringback(/* play the ring if this is the only call*/),
同时发180消息sal_call_notify_ringing(h,propose_early_media || ringback_tone!=NULL);如果自动应答模式,还需接受call.linphone_core_accept_call(lc,call);
具体如下:
【【【---
sal_call_notify_ringing(h,propose_early_media || ringback_tone!=NULL);//180消息
if (propose_early_media || ringback_tone!=NULL){
linphone_call_set_state(call,LinphoneCallIncomingEarlyMedia,"Incoming call early media");
linphone_core_update_streams(lc,call,md);
}
if (sal_call_get_replaces(call->op)!=NULL && lp_config_get_int(lc->config,"sip","auto_answer_replacing_calls",1)){
linphone_core_accept_call(lc,call);
}
---】】】
在换回来说吧,当主叫一方收到180Ring的消息后(case EXOSIP_CALL_RINGING),进入到函数:call_ringing(Sal *sal, eXosip_event_t *ev)
首先:call_proceeding(Sal *sal, eXosip_event_t *ev) --->
/* update contact if received and rport are set by the server
note: will only be used by remote for next INVITE, if any...*/
update_contact_from_response(op,ev->response);
然后得到sdp:sdp=eXosip_get_sdp_info(ev->response);生成本地的mediadesc,然后协商出一个result md,if (op->base.local_media) sdp_process(op);
回调到sip消息回调处理函数里面sal->callbacks.call_ringing(op);,通知界面,开始early medai,ringing了。。。linphone_core_update_streams,即update一下。
又回去,当被叫决定答应呼叫是,他会调用int linphone_core_accept_call(LinphoneCore *lc, LinphoneCall *call);
里面会做:I,/* check if this call is supposed to replace an already running one*/replaced=sal_call_get_replaces(call->op);
II,/*try to be best-effort in giving real local or routable contact address*/ --->sal_op_set_contact(call->op,contact);
III,/*stop ringing */-->ring_stop(lc->ringstream);
IV,if (call->audiostream==NULL)
linphone_call_init_media_streams(call); //这个之前已经做过了,这里只是检查做没做,谨慎期间。
IIV,sal_call_accept(call->op);//发送200OK消息
IIIV,new_md=sal_call_get_final_media_description(call->op);
linphone_core_update_streams(lc, call, new_md); //更新媒体流。。。
VI,通知:ms_message("call answered.");
4,
到这里,一个呼叫的整个常规流程也就完事儿了。而整个linphone的主要功能框架也基本完成。但中间一些细节,如如何添加每个节点的filter,如何维持更新媒体流,借助与mediastream2,怎
么实现流媒体的过程,还需进一步分析。(2011-11-10 宣继托)