From b59ddff3acdb24806a35545f56693798ef952a5f Mon Sep 17 00:00:00 2001
From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
Date: Tue, 22 Sep 2009 17:14:10 -0400
Subject: [PATCH] [#2174] Multiple UDP transports functional (TESTED with 2
accounts and 3 calls)
---
sflphone-common/src/sip/sipvoiplink.cpp | 22 ++++++++++++++++------
1 file changed, 16 insertions(+), 6 deletions(-)
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 535ff631d9..a9aa8bbe3d 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -678,7 +678,7 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
call->setPeerNumber (toUri);
localAddr = account->getSessionAddress ();
- setCallAudioLocal (call, localAddr);
+ setCallAudioLocal (call, call->getLocalIp());
try {
_debug ("Creating new rtp session in newOutgoingCall\n");
@@ -1470,7 +1470,7 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
call->setPeerNumber (toUri);
_debug ("toUri in new_ip_to_ip call %s\n", toUri.c_str());
// Building the local SDP offer
- call->getLocalSDP()->set_ip_address (getLocalIP());
+ call->getLocalSDP()->set_ip_address (getLocalIPAddress ());
call->getLocalSDP()->create_initial_offer();
try {
@@ -1486,9 +1486,9 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
fromUri = account->getFromUri();
- std::string address = findLocalAddressFromUri (toUri, account->getAccountTransport ());
+ std::string address = findLocalAddressFromUri (toUri, _localUDPTransport);
- int port = findLocalPortFromUri (toUri, account->getAccountTransport ());
+ int port = findLocalPortFromUri (toUri, _localUDPTransport);
std::stringstream ss;
@@ -1528,6 +1528,16 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
PJ_ASSERT_RETURN (status == PJ_SUCCESS, false);
+ // Set the appropriate transport
+ pjsip_tpselector *tp;
+ init_transport_selector (_localUDPTransport, &tp);
+ status = pjsip_dlg_set_transport (dialog, tp);
+
+ if (status != PJ_SUCCESS) {
+ _debug ("Failed to set the transport for an IP call\n");
+ return status;
+ }
+
// Associate current call in the invite session
inv->mod_data[getModId() ] = call;
@@ -2460,7 +2470,7 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
_audiortp->stop ();
call->setAudioStart (false);
- _debug ("Create new rtp session from handle_reinvite \n");
+ _debug ("Create new rtp session from handle_reinvite : %s:%i\n", call->getLocalIp().c_str(), call->getLocalAudioPort());
try {
_audiortp->initAudioRtpSession (call);
@@ -2962,7 +2972,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// Have to do some stuff with the SDP
// Set the codec map, IP, peer number and so on... for the SIPCall object
- setCallAudioLocal (call, addrToUse);
+ setCallAudioLocal (call, link->getLocalIPAddress());
// We retrieve the remote sdp offer in the rdata struct to begin the negociation
call->getLocalSDP()->set_ip_address (addrToUse);
--
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