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Unified Diff: webrtc/call/call.h

Issue 2978503002: Move RTP keep-alive config from VideoSendStream::Config to Call::Config (Closed)
Patch Set: Created 3 years, 5 months ago
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Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index 86142f0feaaf065c3215a9e3086bfdf63b08911e..8d8655b3e06ee9363aa4a69b818b8093c6be4c6a 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -112,6 +112,10 @@ class Call {
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* event_log = nullptr;
+
+ // Enables periodic sending if empty keep-alive messages that helps prevent
+ // network time-out events.
pbos-webrtc 2017/07/10 22:40:23 Is there a RFC you can point to here?
sprang_webrtc 2017/07/11 08:39:58 Done.
+ RtpKeepAliveConfig keepalive_config;
};
struct Stats {
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