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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;
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