修改linphone-sdk-android-下篇

前言

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

修改linphone-sdk-android-下篇

前言

问题

分析

总结

guodongAndroid

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