diff --git a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
index 88fdbd1..57c6c05 100644
--- a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
+++ b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
@@ -200,7 +200,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
       LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
       pt_ = i->id;
       audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
-      port2 = rtp_session_get_local_port(audio_stream_->session);
+      port2 = rtp_session_get_local_port(audio_stream_->ms.session);
       first = false;
     }
   }
@@ -211,7 +211,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
     // working with a buggy client; let's try PCMU.
     LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
     audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
-    port2 = rtp_session_get_local_port(audio_stream_->session);
+    port2 = rtp_session_get_local_port(audio_stream_->ms.session);
   }
 
   return true;