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 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 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; inb_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;inumber == 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