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 [外链图片转存失败,源站可能有防盗链机制,建议将图片保存下来直接上传(img-a6dXj5oX-1650980827516)(https://www.iana.org/assignments/_support/sort_none.gif)]
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 INVITEfor (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*/
#endifif (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 availableif (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

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修改linphone-sdk-android-下篇

前言

问题

分析

总结

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