From f23e470d42aeeac99d3af3ed420943275c389cff Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandresavard@alexandresavard-desktop.(none)>
Date: Wed, 24 Mar 2010 13:57:19 -0400
Subject: [PATCH] [#3071] Implement PJSIP's on_create_offer callback
This callback is used when receiving INVITE request without SDP.
---
sflphone-common/src/sip/sipvoiplink.cpp | 51 +++++++++++++++++++------
1 file changed, 39 insertions(+), 12 deletions(-)
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 29c3dbc013..4d068c4862 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -1654,10 +1654,6 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
_debug ("UserAgent: Unable to create RTP Session in new IP2IP call (%s:%d)", __FILE__, __LINE__);
}
- // Building the local SDP offer
- call->getLocalSDP()->set_ip_address (addrSdp);
- call->getLocalSDP()->create_initial_offer(account->getActiveCodecs ());
-
// Init TLS transport if enabled
if(account->isTlsEnabled()) {
@@ -1709,18 +1705,15 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
pj_cstr (&pjTo, toUri.c_str());
pj_str_t pjContact;
-
pj_cstr (&pjContact, contactUri.c_str());
// Create the dialog (UAC)
// (Parameters are "strduped" inside this function)
status = pjsip_dlg_create_uac (pjsip_ua_instance(), &pjFrom, &pjContact, &pjTo, NULL, &dialog);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, false);
// Create the invite session for this call
status = pjsip_inv_create_uac (dialog, call->getLocalSDP()->get_local_sdp_session(), 0, &inv);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, false);
// Set the appropriate transport
@@ -3659,13 +3652,9 @@ mod_on_rx_request (pjsip_rx_data *rdata)
addrSdp = addrToUse;
}
-
call->setConnectionState (Call::Progressing);
-
call->setPeerNumber (peerNumber);
-
call->setDisplayName (displayName);
-
call->initRecFileName();
@@ -4213,7 +4202,45 @@ void on_rx_offer (pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
void on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
- _info("UserAgent: Create new SDP offer ");
+ _info("UserAgent: Create new SDP offer");
+
+ /* Retrieve the call information */
+ SIPCall * call = NULL;
+ call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
+
+ CallID callid = call->getCallId();
+ AccountID accountid = Manager::instance().getAccountFromCall(callid);
+
+ SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accountid));
+
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accountid));
+
+ // Set the local address
+ std::string localAddress = link->getInterfaceAddrFromName(account->getLocalInterface ());
+ // Set SDP parameters - Set to local
+ std::string addrSdp = localAddress;
+
+ _debug ("UserAgent: Local Address for IP2IP call: %s", localAddress.c_str());
+
+ // If local address bound to ANY, reslove it using PJSIP
+ if (localAddress == "0.0.0.0") {
+ link->loadSIPLocalIP (&localAddress);
+ }
+
+ // Local address to appear in SDP
+ if (addrSdp == "0.0.0.0") {
+ addrSdp = localAddress;
+ }
+
+ // Set local address for RTP media
+ setCallAudioLocal (call, localAddress);
+
+ // Building the local SDP offer
+ call->getLocalSDP()->set_ip_address (addrSdp);
+ call->getLocalSDP()->create_initial_offer( account->getActiveCodecs() );
+
+ *p_offer = call->getLocalSDP()->get_local_sdp_session();
+
}
--
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