diff --git a/src/sdp.cpp b/src/sdp.cpp index 63a5b69d58da240126aa7ebe7bc8b140e0660fcd..2c623f8d46e2f9763731c0dc6f07c93ad1927e34 100644 --- a/src/sdp.cpp +++ b/src/sdp.cpp @@ -379,7 +379,6 @@ void Sdp::set_local_media_capabilities () { } } _local_media_cap.push_back (audio); - _debug ("%s\n", audio->to_string ().c_str()); } void Sdp::attribute_port_to_all_media (int port) { @@ -408,7 +407,6 @@ void Sdp::fetch_remote_ip_from_sdp (pjmedia_sdp_session *r_sdp) { std::string remote_ip; remote_ip = r_sdp->conn->addr.ptr; - _debug("************************************************** Remote Audio IP: %s\n", remote_ip.c_str()); this->set_remote_ip(remote_ip); } diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp index f7bd90921c27f63d7e2fcbe17e588e1d55b5b687..b5372eb585757d734988ba9f62c1b1214a97d6c2 100644 --- a/src/sipvoiplink.cpp +++ b/src/sipvoiplink.cpp @@ -1698,14 +1698,12 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam // The call is ringing - We need to handle this case only on outgoing call if (inv->state == PJSIP_INV_STATE_EARLY && e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC){ - _debug ("*************************** PJSIP_INV_STATE_EARLY - PEER RINGING ***********************************\n"); call->setConnectionState(Call::Ringing); Manager::instance().peerRingingCall(call->getCallId()); } // We receive a ACK - The connection is established else if( inv->state == PJSIP_INV_STATE_CONFIRMED ){ - _debug ("*************************** PJSIP_INV_STATE_CONFIRMED ***********************************\n"); /* If the call is a direct IP-to-IP call */ if (call->getCallConfiguration () == Call::IPtoIP) { @@ -1721,7 +1719,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam } else if( inv->state == PJSIP_INV_STATE_DISCONNECTED ){ - _debug ("*************************** PJSIP_INV_STATE_DISCONNECTED %i***********************************\n", inv->cause); switch( inv->cause ) { /* The call terminates normally - BYE / CANCEL */ @@ -2443,8 +2440,6 @@ void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_e void on_rx_offer( pjsip_inv_session *inv, const pjmedia_sdp_session *offer ){ - _debug ( "********************************* REINVITE RECEIVED *******************************\n" ); - #ifdef CAN_REINVITE _debug ("reinvite SIP\n");