diff --git a/pc/channel.cc b/pc/channel.cc
index f83f5cd..8a3a800 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -142,7 +142,7 @@
   RTC_DCHECK_RUN_ON(worker_thread_);
   RTC_DCHECK(ssrc_generator_);
   demuxer_criteria_.mid = content_name;
-  RTC_LOG(LS_INFO) << "Created channel for " << content_name;
+  RTC_LOG(LS_INFO) << "Created channel: " << ToString();
 }
 
 BaseChannel::~BaseChannel() {
@@ -156,12 +156,23 @@
   // the media channel may try to send on the dead transport channel. NULLing
   // is not an effective strategy since the sends will come on another thread.
   media_channel_.reset();
-  RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_;
+  RTC_LOG(LS_INFO) << "Destroyed channel: " << ToString();
+}
+
+std::string BaseChannel::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "{mid: " << content_name_;
+  if (media_channel_) {
+    sb << ", media_type: " << MediaTypeToString(media_channel_->media_type());
+  }
+  sb << "}";
+  return sb.Release();
 }
 
 bool BaseChannel::ConnectToRtpTransport() {
   RTC_DCHECK(rtp_transport_);
   if (!RegisterRtpDemuxerSink()) {
+    RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString();
     return false;
   }
   rtp_transport_->SignalReadyToSend.connect(
@@ -237,7 +248,8 @@
     transport_name_ = rtp_transport_->transport_name();
 
     if (!ConnectToRtpTransport()) {
-      RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport.";
+      RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport for "
+                        << ToString() << ".";
       return false;
     }
     OnTransportReadyToSend(rtp_transport_->IsReadyToSend());
@@ -349,7 +361,7 @@
 
 void BaseChannel::OnNetworkRouteChanged(
     absl::optional<rtc::NetworkRoute> network_route) {
-  RTC_LOG(LS_INFO) << "Network route was changed.";
+  RTC_LOG(LS_INFO) << "Network route for " << ToString() << " was changed.";
 
   RTC_DCHECK(network_thread_->IsCurrent());
   rtc::NetworkRoute new_route;
@@ -404,7 +416,7 @@
 
   // Protect ourselves against crazy data.
   if (!IsValidRtpPacketSize(packet_type, packet->size())) {
-    RTC_LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
+    RTC_LOG(LS_ERROR) << "Dropping outgoing " << ToString() << " "
                       << RtpPacketTypeToString(packet_type)
                       << " packet: wrong size=" << packet->size();
     return false;
@@ -420,16 +432,16 @@
       }
       // However, there shouldn't be any RTP packets sent before SRTP is set up
       // (and SetSend(true) is called).
-      RTC_LOG(LS_ERROR)
-          << "Can't send outgoing RTP packet when SRTP is inactive"
-             " and crypto is required";
+      RTC_LOG(LS_ERROR) << "Can't send outgoing RTP packet for " << ToString()
+                        << " when SRTP is inactive and crypto is required";
       RTC_NOTREACHED();
       return false;
     }
 
     std::string packet_type = rtcp ? "RTCP" : "RTP";
     RTC_LOG(LS_WARNING) << "Sending an " << packet_type
-                        << " packet without encryption.";
+                        << " packet without encryption for " << ToString()
+                        << ".";
   }
 
   // Bon voyage.
@@ -463,7 +475,8 @@
     //    for us to just eat packets here. This is all sidestepped if RTCP mux
     //    is used anyway.
     RTC_LOG(LS_WARNING) << "Can't process incoming RTP packet when "
-                           "SRTP is inactive and crypto is required";
+                           "SRTP is inactive and crypto is required "
+                        << ToString();
     return;
   }
 
@@ -504,7 +517,7 @@
   if (enabled_)
     return;
 
-  RTC_LOG(LS_INFO) << "Channel enabled";
+  RTC_LOG(LS_INFO) << "Channel enabled: " << ToString();
   enabled_ = true;
   UpdateMediaSendRecvState_w();
 }
@@ -514,7 +527,7 @@
   if (!enabled_)
     return;
 
-  RTC_LOG(LS_INFO) << "Channel disabled";
+  RTC_LOG(LS_INFO) << "Channel disabled: " << ToString();
   enabled_ = false;
   UpdateMediaSendRecvState_w();
 }
@@ -534,7 +547,7 @@
     return;
   }
 
-  RTC_LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
+  RTC_LOG(LS_INFO) << "Channel writable (" << ToString() << ")"
                    << (was_ever_writable_ ? "" : " for the first time");
 
   was_ever_writable_ = true;
@@ -547,7 +560,7 @@
   if (!writable_)
     return;
 
-  RTC_LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
+  RTC_LOG(LS_INFO) << "Channel not writable (" << ToString() << ")";
   writable_ = false;
   UpdateMediaSendRecvState();
 }
@@ -632,7 +645,8 @@
     }
 
     if (media_channel()->AddSendStream(new_stream)) {
-      RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0];
+      RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0]
+                       << " into " << ToString();
     } else {
       rtc::StringBuilder desc;
       desc << "Failed to add send stream ssrc: " << new_stream.first_ssrc();
@@ -655,11 +669,13 @@
     // the unsignaled stream params that are cached.
     if (!old_stream.has_ssrcs() && !HasStreamWithNoSsrcs(streams)) {
       ResetUnsignaledRecvStream_w();
-      RTC_LOG(LS_INFO) << "Reset unsignaled remote stream.";
+      RTC_LOG(LS_INFO) << "Reset unsignaled remote stream for " << ToString()
+                       << ".";
     } else if (old_stream.has_ssrcs() &&
                !GetStreamBySsrc(streams, old_stream.first_ssrc())) {
       if (RemoveRecvStream_w(old_stream.first_ssrc())) {
-        RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc();
+        RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc()
+                         << " from " << ToString() << ".";
       } else {
         rtc::StringBuilder desc;
         desc << "Failed to remove remote stream with ssrc "
@@ -681,13 +697,15 @@
         RTC_LOG(LS_INFO) << "Add remote ssrc: "
                          << (new_stream.has_ssrcs()
                                  ? std::to_string(new_stream.first_ssrc())
-                                 : "unsignaled");
+                                 : "unsignaled")
+                         << " to " << ToString();
       } else {
         rtc::StringBuilder desc;
         desc << "Failed to add remote stream ssrc: "
              << (new_stream.has_ssrcs()
                      ? std::to_string(new_stream.first_ssrc())
-                     : "unsignaled");
+                     : "unsignaled")
+             << " to " << ToString();
         SafeSetError(desc.str(), error_desc);
         ret = false;
       }
@@ -697,7 +715,9 @@
                                    new_stream.ssrcs.end());
   }
   // Re-register the sink to update the receiving ssrcs.
-  RegisterRtpDemuxerSink();
+  if (!RegisterRtpDemuxerSink()) {
+    RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString();
+  }
   remote_streams_ = streams;
   return ret;
 }
@@ -813,7 +833,8 @@
   bool send = IsReadyToSendMedia_w();
   media_channel()->SetSend(send);
 
-  RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
+  RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send
+                   << " for " << ToString();
 }
 
 bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
@@ -821,7 +842,7 @@
                                      std::string* error_desc) {
   TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting local voice description";
+  RTC_LOG(LS_INFO) << "Setting local voice description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -852,7 +873,7 @@
     }
     // Need to re-register the sink to update the handled payload.
     if (!RegisterRtpDemuxerSink()) {
-      RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing.";
+      RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing for " << ToString();
       return false;
     }
   }
@@ -878,7 +899,7 @@
                                       std::string* error_desc) {
   TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting remote voice description";
+  RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -907,10 +928,11 @@
 
   if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) {
     RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - "
-                            "disable payload type demuxing";
+                            "disable payload type demuxing for "
+                         << ToString();
     ClearHandledPayloadTypes();
     if (!RegisterRtpDemuxerSink()) {
-      RTC_LOG(LS_ERROR) << "Failed to update audio demuxing.";
+      RTC_LOG(LS_ERROR) << "Failed to update audio demuxing for " << ToString();
       return false;
     }
   }
@@ -958,11 +980,12 @@
   // and we have had some form of connectivity.
   bool send = IsReadyToSendMedia_w();
   if (!media_channel()->SetSend(send)) {
-    RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel";
+    RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel: " + ToString();
     // TODO(gangji): Report error back to server.
   }
 
-  RTC_LOG(LS_INFO) << "Changing video state, send=" << send;
+  RTC_LOG(LS_INFO) << "Changing video state, send=" << send << " for "
+                   << ToString();
 }
 
 void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
@@ -975,7 +998,7 @@
                                      std::string* error_desc) {
   TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting local video description";
+  RTC_LOG(LS_INFO) << "Setting local video description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -1027,7 +1050,7 @@
     }
     // Need to re-register the sink to update the handled payload.
     if (!RegisterRtpDemuxerSink()) {
-      RTC_LOG(LS_ERROR) << "Failed to set up video demuxing.";
+      RTC_LOG(LS_ERROR) << "Failed to set up video demuxing for " << ToString();
       return false;
     }
   }
@@ -1061,7 +1084,7 @@
                                       std::string* error_desc) {
   TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting remote video description";
+  RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -1120,10 +1143,11 @@
 
   if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) {
     RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - "
-                            "disable payload type demuxing";
+                            "disable payload type demuxing for "
+                         << ToString();
     ClearHandledPayloadTypes();
     if (!RegisterRtpDemuxerSink()) {
-      RTC_LOG(LS_ERROR) << "Failed to update video demuxing.";
+      RTC_LOG(LS_ERROR) << "Failed to update video demuxing for " << ToString();
       return false;
     }
   }
@@ -1203,7 +1227,7 @@
                                        std::string* error_desc) {
   TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting local data description";
+  RTC_LOG(LS_INFO) << "Setting local data description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -1233,7 +1257,7 @@
   }
   // Need to re-register the sink to update the handled payload.
   if (!RegisterRtpDemuxerSink()) {
-    RTC_LOG(LS_ERROR) << "Failed to set up data demuxing.";
+    RTC_LOG(LS_ERROR) << "Failed to set up data demuxing for " << ToString();
     return false;
   }
 
@@ -1258,7 +1282,7 @@
                                         std::string* error_desc) {
   TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w");
   RTC_DCHECK_RUN_ON(worker_thread());
-  RTC_LOG(LS_INFO) << "Setting remote data description";
+  RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString();
 
   RTC_DCHECK(content);
   if (!content) {
@@ -1280,7 +1304,7 @@
   RtpHeaderExtensions rtp_header_extensions =
       GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
 
-  RTC_LOG(LS_INFO) << "Setting remote data description";
+  RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString();
   DataSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription<DataCodec>(
       data, rtp_header_extensions,
@@ -1311,20 +1335,21 @@
   // content. We receive data on the default channel and multiplexed streams.
   bool recv = IsReadyToReceiveMedia_w();
   if (!media_channel()->SetReceive(recv)) {
-    RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel";
+    RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel: " << ToString();
   }
 
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
   bool send = IsReadyToSendMedia_w();
   if (!media_channel()->SetSend(send)) {
-    RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel";
+    RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel: " << ToString();
   }
 
   // Trigger SignalReadyToSendData asynchronously.
   OnDataChannelReadyToSend(send);
 
-  RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
+  RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send
+                   << " for " << ToString();
 }
 
 void RtpDataChannel::OnMessage(rtc::Message* pmsg) {
diff --git a/pc/channel.h b/pc/channel.h
index 238a8e2..9dc6522 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -275,6 +275,9 @@
 
   bool RegisterRtpDemuxerSink();
 
+  // Return description of media channel to facilitate logging
+  std::string ToString() const;
+
   bool has_received_packet_ = false;
 
  private: