RTP: Send silence packets on idle streams for every second.
Originally a stream does not send packets when it is receive-only or there is
nothing to mix. However, this causes some problems with certain firewalls and
proxies. A firewall might remove a port mapping when there is no outgoing
packet for a preiod of time, and a proxy might wait for incoming packets from
both sides before start forwarding. To solve these problems, we send out a
silence packet on the stream for every second. It should be good enough to
keep the stream alive with relatively low resources.
Bug: 3119690
Change-Id: Ib9c55e5dddfba28928bd9b376832b68bda24c0e4
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 0c8a725..60abf2a 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -63,6 +63,14 @@
// real jitter buffer. For a stream at 8000Hz it takes 8192 bytes. These numbers
// are chosen by experiments and each of them can be adjusted as needed.
+// Originally a stream does not send packets when it is receive-only or there is
+// nothing to mix. However, this causes some problems with certain firewalls and
+// proxies. A firewall might remove a port mapping when there is no outgoing
+// packet for a preiod of time, and a proxy might wait for incoming packets from
+// both sides before start forwarding. To solve these problems, we send out a
+// silence packet on the stream for every second. It should be good enough to
+// keep the stream alive with relatively low resources.
+
// Other notes:
// + We use elapsedRealtime() to get the time. Since we use 32bit variables
// instead of 64bit ones, comparison must be done by subtraction.
@@ -110,7 +118,7 @@
int mSampleRate;
int mSampleCount;
int mInterval;
- int mLogThrottle;
+ int mKeepAlive;
int16_t *mBuffer;
int mBufferMask;
@@ -262,12 +270,8 @@
++mSequence;
mTimestamp += mSampleCount;
- if (mMode == RECEIVE_ONLY) {
- return;
- }
-
// If there is an ongoing DTMF event, send it now.
- if (mDtmfEvent != -1) {
+ if (mMode != RECEIVE_ONLY && mDtmfEvent != -1) {
int duration = mTimestamp - mDtmfStart;
// Make sure duration is reasonable.
if (duration >= 0 && duration < mSampleRate * 100) {
@@ -289,43 +293,55 @@
mDtmfEvent = -1;
}
- // It is time to mix streams.
- bool mixed = false;
int32_t buffer[mSampleCount + 3];
- memset(buffer, 0, sizeof(buffer));
- while (chain) {
- if (chain != this &&
- chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
- mixed = true;
+ int16_t samples[mSampleCount];
+ if (mMode == RECEIVE_ONLY) {
+ if ((mTick ^ mKeepAlive) >> 10 == 0) {
+ return;
}
- chain = chain->mNext;
- }
- if (!mixed) {
- if ((mTick ^ mLogThrottle) >> 10) {
- mLogThrottle = mTick;
+ mKeepAlive = mTick;
+ memset(samples, 0, sizeof(samples));
+ } else {
+ // Mix all other streams.
+ bool mixed = false;
+ memset(buffer, 0, sizeof(buffer));
+ while (chain) {
+ if (chain != this &&
+ chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
+ mixed = true;
+ }
+ chain = chain->mNext;
+ }
+
+ if (mixed) {
+ // Saturate into 16 bits.
+ for (int i = 0; i < mSampleCount; ++i) {
+ int32_t sample = buffer[i];
+ if (sample < -32768) {
+ sample = -32768;
+ }
+ if (sample > 32767) {
+ sample = 32767;
+ }
+ samples[i] = sample;
+ }
+ } else {
+ if ((mTick ^ mKeepAlive) >> 10 == 0) {
+ return;
+ }
+ mKeepAlive = mTick;
+ memset(samples, 0, sizeof(samples));
LOGV("stream[%d] no data", mSocket);
}
- return;
}
- // Cook the packet and send it out.
- int16_t samples[mSampleCount];
- for (int i = 0; i < mSampleCount; ++i) {
- int32_t sample = buffer[i];
- if (sample < -32768) {
- sample = -32768;
- }
- if (sample > 32767) {
- sample = 32767;
- }
- samples[i] = sample;
- }
if (!mCodec) {
// Special case for device stream.
send(mSocket, samples, sizeof(samples), MSG_DONTWAIT);
return;
}
+ // Cook the packet and send it out.
buffer[0] = htonl(mCodecMagic | mSequence);
buffer[1] = htonl(mTimestamp);
buffer[2] = mSsrc;
@@ -883,7 +899,7 @@
int codecType = -1;
char codecName[16];
int sampleRate = -1;
- sscanf(codecSpec, "%d %[^/]%*c%d", &codecType, codecName, &sampleRate);
+ sscanf(codecSpec, "%d %15[^/]%*c%d", &codecType, codecName, &sampleRate);
codec = newAudioCodec(codecName);
int sampleCount = (codec ? codec->set(sampleRate, codecSpec) : -1);
env->ReleaseStringUTFChars(jCodecSpec, codecSpec);
