修改linphone-sdk-android-下篇

前言

接上篇修改linphone-sdk-android-上篇

接中篇修改linphone-sdk-android-中篇

本文是下篇，本篇记录在上篇中提到的问题1排查过程及修复方案，尽量描述排查问题过程中的思路与方向

上篇中说问题1当初认为是linphone的bug，后面看源码及查资料发现可能不是bug，本篇将记录个人的理解

问题

这里再描述下问题1：打开音频编解码G722、G729等时，发起呼叫的INVITE SDP中，没有G722、G729的rtpmap

m=audio 7078 RTP/AVP 96 0 8 9 18 101 97
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:97 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

分析

这里先了解下SDP协议，参考The Session Description Protocol (SDP) (3cx.com)

rtpmap是Session attribute lines，即为会话属性行，是对Payload Type的补充说明，Payload Type既是m=audio 7078 RTP/AVP 96 0 8 9 18 101 97中AVP后面的数字，这些数字是音频编解码对应的代码，对应关系如下：

下表源自Real-Time Transport Protocol (RTP) Parameters (iana.org)

PT	Encoding Name	Audio/Video (A/V)	Clock Rate (Hz)	Channels	Reference
0	PCMU	A	8000	1	[RFC3551]
1	Reserved
2	Reserved
3	GSM	A	8000	1	[RFC3551]
4	G723	A	8000	1	[Vineet_Kumar][RFC3551]
5	DVI4	A	8000	1	[RFC3551]
6	DVI4	A	16000	1	[RFC3551]
7	LPC	A	8000	1	[RFC3551]
8	PCMA	A	8000	1	[RFC3551]
9	G722	A	8000	1	[RFC3551]
10	L16	A	44100	2	[RFC3551]
11	L16	A	44100	1	[RFC3551]
12	QCELP	A	8000	1	[RFC3551]
13	CN	A	8000	1	[RFC3389]
14	MPA	A	90000		[RFC3551][RFC2250]
15	G728	A	8000	1	[RFC3551]
16	DVI4	A	11025	1	[Joseph_Di_Pol]
17	DVI4	A	22050	1	[Joseph_Di_Pol]
18	G729	A	8000	1	[RFC3551]
19	Reserved	A
20	Unassigned	A
21	Unassigned	A
22	Unassigned	A
23	Unassigned	A
24	Unassigned	V
25	CelB	V	90000		[RFC2029]
26	JPEG	V	90000		[RFC2435]
27	Unassigned	V
28	nv	V	90000		[RFC3551]
29	Unassigned	V
30	Unassigned	V
31	H261	V	90000		[RFC4587]
32	MPV	V	90000		[RFC2250]
33	MP2T	AV	90000		[RFC2250]
34	H263	V	90000		[Chunrong_Zhu]
35-71	Unassigned	?
72-76	Reserved for RTCP conflict avoidance				[RFC3551]
77-95	Unassigned	?
96-127	dynamic	?			[RFC3551]

从表中了解到，Payload Type(PT) code 0 - 95为静态类型，即code对应固定的codec(编解码器)，96 - 127为动态codec，即需要在SDP协商过程中确定

接下来追踪下源码，看看SDP中为什么没有rtpmap

先找到Java层发起呼叫的代码，在Core.java中有4个发起呼叫的方法

@Nullable
Call invite(@NonNull String var1);

@Nullable
Call inviteAddress(@NonNull Address var1);

@Nullable
Call inviteAddressWithParams(@NonNull Address var1, @NonNull CallParams var2);

@Nullable
Call inviteWithParams(@NonNull String var1, @NonNull CallParams var2);

具体实现在CoreImpl.java中，查看这个public Call inviteAddress(@NonNull Address addr);方法吧

private native Call inviteAddress(long nativePtr, Address addr);

@Override @Nullable
synchronized public Call inviteAddress(@NonNull Address addr)  {
    return (Call)inviteAddress(nativePtr, addr);
}

Java层调用了native层代码，打开编译后生成的linphone_jni.cc，找到Java_org_linphone_core_CoreImpl_inviteAddress方法

JNIEXPORT jobject JNICALL Java_org_linphone_core_CoreImpl_inviteAddress(JNIEnv *env, jobject thiz, jlong ptr, jobject addr) {
	LinphoneCore *cptr = (LinphoneCore*)ptr;
	if (cptr == nullptr) {
		bctbx_error("Java_org_linphone_core_CoreImpl_inviteAddress's LinphoneCore C ptr is null!");
		return 0;
	}
	
	LinphoneAddress* c_addr = nullptr;
	if (addr) c_addr = (LinphoneAddress*)GetObjectNativePtr(env, addr);
	
	jobject jni_result = (jobject)getCall(env, (LinphoneCall *)linphone_core_invite_address(cptr, c_addr), TRUE);
	return jni_result;
}

在native层调用了linphone_core_invite_address这个方法，在IDE中，可以通过Ctrl+左键点击进行跳转，linphone_core_invite_address位于linphonecore.c中

LinphoneCall * linphone_core_invite_address(LinphoneCore *lc, const LinphoneAddress *addr){
	LinphoneCall *call;
	LinphoneCallParams *p=linphone_core_create_call_params(lc, NULL);
	linphone_call_params_enable_video(p, linphone_call_params_video_enabled(p) && !!lc->video_policy.automatically_initiate);
	call=linphone_core_invite_address_with_params (lc,addr,p);
	linphone_call_params_unref(p);
	return call;
}

在linphone_core_invite_address方法中调用了linphone_core_invite_address_with_params发起呼叫，这个方法较长，删减一些不关心的代码

LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const LinphoneAddress *addr, const LinphoneCallParams *params){
	const char *from=NULL;
	LinphoneCall *call;
    
	if (!addr) {
		ms_error("Can't invite a NULL address");
		return NULL;
	}

	parsed_url2=linphone_address_new(from);
	call=linphone_call_new_outgoing(lc,parsed_url2,addr,cp,proxy);
    
    bool defer = Call::toCpp(call)->initiateOutgoing();
	if (!defer) {
		if (Call::toCpp(call)->startInvite(nullptr) != 0) {
			/* The call has already gone to error and released state, so do not return it */
			call = nullptr;
		}
	}

	return call;
}

在linphone_core_invite_address_with_params方法中调用linphone_call_new_outgoing方法创建Call对象，调用initiateOutgoing方法初始化发起呼叫并设置当前状态为OutgoingInit，接下来调用startInvite方法发起呼叫，startInvite方法位于call.cpp中，在其中又调用getActiveSession方法获取CallSession，调用CallSession::startInvite方法

int Call::startInvite (const Address *destination) {
	return getActiveSession()->startInvite(destination, "");
}

CallSession::startInvite方法位于call-session.cpp中，在这个方法中找了半天，没见有与SDP发送相关的逻辑，先去头文件中看看方法原型吧

找了半天也是有点收获的，分析出调用addAdditionalLocalBody去组装自定义扩展头数据

int CallSession::startInvite (const Address *destination, const string &subject, const Content *content) {
	L_D();
	d->subject = subject;
	/* Try to be best-effort in giving real local or routable contact address */
	d->setContactOp();
	string destinationStr;
	char *realUrl = nullptr;
	if (destination)
		destinationStr = destination->asString();
	else {
		realUrl = linphone_address_as_string(d->log->to);
		destinationStr = realUrl;
		ms_free(realUrl);
	}
	char *from = linphone_address_as_string(d->log->from);
	/* Take a ref because sal_call() may destroy the CallSession if no SIP transport is available */
	shared_ptr<CallSession> ref = getSharedFromThis();
	if (content)
		d->op->setLocalBody(*content);

	// If a custom Content has been set in the call params, create a multipart body for the INVITE
	for (auto& content : d->params->getCustomContents()) {
		d->op->addAdditionalLocalBody(content);
	}

	int result = d->op->call(from, destinationStr, subject);
	ms_free(from);
	if (result < 0) {
		if ((d->state != CallSession::State::Error) && (d->state != CallSession::State::Released)) {
			// sal_call() may invoke call_failure() and call_released() SAL callbacks synchronously,
			// in which case there is no need to perform a state change here.
			d->setState(CallSession::State::Error, "Call failed");
		}
	} else {
		linphone_call_log_set_call_id(d->log, d->op->getCallId().c_str()); /* Must be known at that time */
		d->setState(CallSession::State::OutgoingProgress, "Outgoing call in progress");
	}
	return result;
}

CallSession::startInvite方法原型为，

virtual int startInvite (const Address *destination, const std::string &subject = "", const Content *content = nullptr);

是个virtual虚函数，说明有函数复写，在IDE中搜索发现MediaSession类继承自CallSession，好的，找到MediaSession复写的startInvite方法，方法较长，删除一些不关心的代码

int MediaSession::startInvite (const Address *destination, const string &subject, const Content *content) {
	L_D();
	
	// 删除不关心的代码

	d->op->setLocalMediaDescription(d->localDesc);

	int result = CallSession::startInvite(destination, subject, content);
	if (result < 0) {
		if (d->state == CallSession::State::Error)
			d->stopStreams();
		return result;
	}
	return result;
}

在MediaSession::startInvite中调用setLocalMediaDescription方法组装本地媒体描述信息，最后再调用父类的CallSession::startInvite方法继续发起呼叫，好的，现在只关心setLocalMediaDescription方法，其中op是SalCallOp，在IDE中打开call-op.cpp，找到setLocalMediaDescription方法，删减一些不关心的代码

int SalCallOp::setLocalMediaDescription (SalMediaDescription *desc) {
	if (desc) {
		sal_media_description_ref(desc);
		belle_sip_error_code error;
		belle_sdp_session_description_t *sdp = media_description_to_sdp(desc);
		vector<char> buffer = marshalMediaDescription(sdp, error);
		belle_sip_object_unref(sdp);
		if (error != BELLE_SIP_OK)
			return -1;

		mLocalBody.setContentType(ContentType::Sdp);
		mLocalBody.setBody(move(buffer));
	} else {
		mLocalBody = Content();
	}
	return 0;
}

到这里终于发现与SDP相关的方法了media_description_to_sdp，继续查看media_description_to_sdp方法，此方法位于sal_sdp.c中，方法较长，主要是组装SDP协议数据，比如设置版本、创建源信息，创建会话等，这里删减一些不关心的代码

belle_sdp_session_description_t * media_description_to_sdp(const SalMediaDescription *desc) {
	belle_sdp_session_description_t* session_desc=belle_sdp_session_description_new();
	bool_t inet6;
	belle_sdp_origin_t* origin;
	int i;
	char *escaped_username = belle_sip_uri_to_escaped_username(desc->username);

	if ( strchr ( desc->addr,':' ) !=NULL ) {
		inet6=1;
	} else inet6=0;
	belle_sdp_session_description_set_version ( session_desc,belle_sdp_version_create ( 0 ) );

	origin = belle_sdp_origin_create ( escaped_username
									  ,desc->session_id
									  ,desc->session_ver
									  ,"IN"
									  , inet6 ? "IP6" :"IP4"
									  ,desc->addr );
	bctbx_free(escaped_username);

	belle_sdp_session_description_set_origin ( session_desc,origin );

	belle_sdp_session_description_set_session_name ( session_desc,
		belle_sdp_session_name_create ( desc->name[0]!='\0' ? desc->name : "Talk" ) );

	// 删减不关心的代码

	for ( i=0; i<desc->nb_streams; i++ ) {
		stream_description_to_sdp(session_desc, desc, &desc->streams[i]);
	}
	return session_desc;
}

分析media_description_to_sdp方法找到在stream_description_to_sdp方法中组装数据流信息到SDP协议中，stream_description_to_sdp方法非常长，此方法主要是组装SDP协议中编解码相关的信息，这里删除大部分不关心的代码

static void stream_description_to_sdp ( belle_sdp_session_description_t *session_desc, const SalMediaDescription *md, const SalStreamDescription *stream ) {
    
    // 删减不关心的代码

	media_desc = belle_sdp_media_description_create ( sal_stream_description_get_type_as_string(stream)
				 ,stream->rtp_port
				 ,1
				 ,sal_media_proto_to_string ( stream->proto )
				 ,NULL );
    // 看到payloads字段
	if (stream->payloads) {
		for ( pt_it=stream->payloads; pt_it!=NULL; pt_it=pt_it->next ) {
			pt= ( PayloadType* ) pt_it->data;
			mime_param= belle_sdp_mime_parameter_create ( pt->mime_type
					, payload_type_get_number ( pt )
					, pt->clock_rate
					, pt->channels>0 ? pt->channels : -1 );
			belle_sdp_mime_parameter_set_parameters ( mime_param,pt->recv_fmtp );
			if ( stream->ptime>0 ) {
				belle_sdp_mime_parameter_set_ptime ( mime_param,stream->ptime );
			}
            // 锁定此方法
			belle_sdp_media_description_append_values_from_mime_parameter ( media_desc,mime_param );
			belle_sip_object_unref ( mime_param );
		}
	} else {
		/* to comply with SDP we cannot have an empty payload type number list */
		/* as it happens only when mline is declined with a zero port, it does not matter to put whatever codec*/
		belle_sip_list_t* format = belle_sip_list_append(NULL,0);
		belle_sdp_media_set_media_formats(belle_sdp_media_description_get_media(media_desc),format);
	}

    // 组装自定义sdp属性
	if (stream->custom_sdp_attributes) {
		belle_sdp_session_description_t *custom_desc = (belle_sdp_session_description_t *)stream->custom_sdp_attributes;
		belle_sip_list_t *l = belle_sdp_session_description_get_attributes(custom_desc);
		belle_sip_list_t *elem;
		for (elem = l; elem != NULL; elem = elem->next) {
			belle_sdp_media_description_add_attribute(media_desc, (belle_sdp_attribute_t *)elem->data);
		}
	}
    
    // 删减不关心的代码
}

在stream_description_to_sdp方法中看到payload字段，马上就要找到了happy~

经过分析，锁定belle_sdp_media_description_append_values_from_mime_parameter方法，分析此方法，在其中找到组装rtpmap的源码

void belle_sdp_media_description_append_values_from_mime_parameter(belle_sdp_media_description_t* media_description, const belle_sdp_mime_parameter_t* mime_parameter) {
    
#ifndef BELLE_SDP_FORCE_RTP_MAP /* defined to for RTP map even for static codec*/
	if (!mime_parameter_is_static(mime_parameter)) {
		/*dynamic payload*/
#endif
		if (belle_sdp_mime_parameter_get_channel_count(mime_parameter)>1) {
			snprintf(atribute_value,MAX_FMTP_LENGTH,"%i %s/%i/%i"
					,belle_sdp_mime_parameter_get_media_format(mime_parameter)
					,belle_sdp_mime_parameter_get_type(mime_parameter)
					,belle_sdp_mime_parameter_get_rate(mime_parameter)
					,belle_sdp_mime_parameter_get_channel_count(mime_parameter));
		} else {
			snprintf(atribute_value,MAX_FMTP_LENGTH,"%i %s/%i"
					,belle_sdp_mime_parameter_get_media_format(mime_parameter)
					,belle_sdp_mime_parameter_get_type(mime_parameter)
					,belle_sdp_mime_parameter_get_rate(mime_parameter));
		}
		belle_sdp_media_description_set_attribute_value(media_description,"rtpmap",atribute_value);
#ifndef BELLE_SDP_FORCE_RTP_MAP
	}
#endif
    
    // always include fmtp parameters if available
	if (belle_sdp_mime_parameter_get_parameters(mime_parameter)) {
		snprintf(atribute_value,MAX_FMTP_LENGTH,"%i %s"
				,belle_sdp_mime_parameter_get_media_format(mime_parameter)
				,belle_sdp_mime_parameter_get_parameters(mime_parameter));
		belle_sdp_media_description_set_attribute_value(media_description,"fmtp",atribute_value);
	}
}

这里先分析下mime_parameter_is_static方法是干什么的？查看以下源码发现，噢~~，原来是用于判断编解码是否是静态类型(前面提到的Payload Type)

const struct static_payload static_payload_list [] ={
	/*audio*/
	{0,1,"PCMU",8000},
	{3,1,"GSM",8000},
	{4,1,"G723",8000},
	{5,1,"DVI4",8000},
	{6,1,"DVI4",16000},
	{7,1,"LPC",8000},
	{8,1,"PCMA",8000},
	{9,1,"G722",8000},
	{10,2,"L16",44100},
	{11,1,"L16",44100},
	{12,1,"QCELP",8000},
	{13,1,"CN",8000},
	{14,1,"MPA",90000},
	{15,1,"G728",8000},
	{16,1,"DVI4",11025},
	{17,1,"DVI4",22050},
	{18,1,"G729",8000},
	/*video*/
	{25,0,"CelB",90000},
	{26,0,"JPEG",90000},
	{28,0,"nv",90000},
	{31,0,"H261",90000},
	{32,0,"MPV",90000},
	{33,0,"MP2T",90000},
	{34,0,"H263",90000}
};

static int mime_parameter_is_static(const belle_sdp_mime_parameter_t *param){
	const struct static_payload* iterator;
	size_t i;

	for (iterator = static_payload_list,i=0;i<payload_list_elements;i++,iterator++) {
		if (iterator->number == param->media_format &&
			strcasecmp(iterator->type,param->type)==0 &&
			iterator->channel_count==param->channel_count &&
			iterator->rate==param->rate ) {
			return TRUE;
		}
	}
	return FALSE;
}

现在再来分析下belle_sdp_media_description_append_values_from_mime_parameter方法的意思，大意如下：如果没有定义BELLE_SDP_FORCE_RTP_MAP这个宏就执行if (!mime_parameter_is_static(mime_parameter))判断编解码是否是静态类型，如果定义了就不判断是否是静态类型

总结一下就是如果没有定义BELLE_SDP_FORCE_RTP_MAP这个宏，就不组装静态类型编解码的rtpmap信息，只组装动态类型编解码的rtpmap信息，终于找到源头了，真是拨云见日呀

到这里还没完，既然是根据宏定义做的判断，肯定在编译的时候可以配置，先看看能不能找到定义宏的地方，在IDE中全局搜索，在belle-sip下的CMakeList.txt中发现

option(ENABLE_RTP_MAP_ALWAYS_IN_SDP "Always include rtpmap in SDP." OFF)

if(ENABLE_RTP_MAP_ALWAYS_IN_SDP) 
	set(BELLE_SDP_FORCE_RTP_MAP 1)
endif()

bingo~，真的是到最后了

最后在编译时增加编译配置项

$ cd linphone-sdk/build/
$ cmake -DENABLE_RTP_MAP_ALWAYS_IN_SDP=ON ..
$ cmake --build . --parallel 8

重新编译后拷贝到AS中运行，发起呼叫查看Logcat输出

总结

在源码中看到通过BELLE_SDP_FORCE_RTP_MAP这个宏控制是否在SDP中包含静态类型编解码的rtpmap信息，个人猜测是静态类型的编解码信息，是协议中固定的，任何遵循协议的实现方，都可以根据静态类型编解码对应的code解析出相应的rtpmap信息，所以在SDP中去掉静态类型编解码器的rtpmap信息，同时也可以减少发送数据包的大小，减轻网络压力