From c3fdb98f07e2ae5223a4780b025f698fe6fb7154 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandresavard@alexandresavard-desktop.(none)>
Date: Thu, 25 Mar 2010 15:06:52 -0400
Subject: [PATCH] [#3071] Fix segfault on receiving INVITE without SDP, send
488 error
---
sflphone-common/src/sip/sdp.cpp | 23 +++++++++++++++
sflphone-common/src/sip/sdp.h | 19 ++++++-------
sflphone-common/src/sip/sipvoiplink.cpp | 19 +++----------
sflphone-common/src/sip/sipvoiplink.h | 38 ++++++++++++-------------
4 files changed, 54 insertions(+), 45 deletions(-)
diff --git a/sflphone-common/src/sip/sdp.cpp b/sflphone-common/src/sip/sdp.cpp
index 94c8e64634..5b3066212a 100644
--- a/sflphone-common/src/sip/sdp.cpp
+++ b/sflphone-common/src/sip/sdp.cpp
@@ -211,6 +211,10 @@ int Sdp::receiving_initial_offer (pjmedia_sdp_session* remote, CodecOrder select
pj_status_t status;
+ if (!remote) {
+ return !PJ_SUCCESS;
+ }
+
// Create the SDP negociator instance by calling
// pjmedia_sdp_neg_create_w_remote_offer with the remote offer, and by providing the local offer ( optional )
@@ -531,6 +535,19 @@ AudioCodec* Sdp::get_session_media (void)
}
+pj_status_t Sdp::start_negociation()
+{
+ pj_status_t status;
+
+ if (_negociator) {
+ status = pjmedia_sdp_neg_negotiate(_pool, _negociator, 0);
+ }
+ else {
+ status = !PJ_SUCCESS;
+ }
+
+ return status;
+}
void Sdp::toString (void)
{
@@ -648,6 +665,9 @@ void Sdp::set_media_transport_info_from_remote_sdp (const pjmedia_sdp_session *r
_info ("SDP: Fetching media from sdp");
+ if(!remote_sdp)
+ return;
+
pjmedia_sdp_media *r_media;
this->get_remote_sdp_media_from_offer (remote_sdp, &r_media);
@@ -667,6 +687,9 @@ void Sdp::get_remote_sdp_media_from_offer (const pjmedia_sdp_session* remote_sdp
{
int count, i;
+ if(!remote_sdp)
+ return;
+
count = remote_sdp->media_count;
*r_media = NULL;
diff --git a/sflphone-common/src/sip/sdp.h b/sflphone-common/src/sip/sdp.h
index e3932751aa..8c81168adf 100644
--- a/sflphone-common/src/sip/sdp.h
+++ b/sflphone-common/src/sip/sdp.h
@@ -166,10 +166,7 @@ class Sdp {
* @return pj_status_t 0 on success
* 1 otherwise
*/
- pj_status_t start_negociation( void ){
- return pjmedia_sdp_neg_negotiate(
- _pool, _negociator, 0);
- }
+ pj_status_t start_negociation( void );
/*
* Retrieve the negociated sdp offer from the sip payload.
@@ -221,7 +218,7 @@ class Sdp {
std::vector<sdpMedia*> get_session_media_list (void) { return _session_media; }
- void get_remote_sdp_crypto_from_offer (const pjmedia_sdp_session* remote_sdp, CryptoOffer& crypto_offer);
+ void get_remote_sdp_crypto_from_offer (const pjmedia_sdp_session* remote_sdp, CryptoOffer& crypto_offer);
private:
/** Codec Map */
@@ -253,13 +250,13 @@ class Sdp {
/** Local audio port */
int _local_extern_audio_port;
- /** Remote's audio port */
+ /** Remote audio port */
unsigned int _remote_audio_port;
std::string _zrtp_hello_hash;
- /** "a=crypto" sdes local attributes obtained from AudioSrtpSession */
- std::vector<std::string> _srtp_crypto;
+ /** "a=crypto" sdes local attributes obtained from AudioSrtpSession */
+ std::vector<std::string> _srtp_crypto;
Sdp(const Sdp&); //No Copy Constructor
Sdp& operator=(const Sdp&); //No Assignment Operator
@@ -350,12 +347,12 @@ class Sdp {
void get_remote_sdp_media_from_offer (const pjmedia_sdp_session* r_sdp, pjmedia_sdp_media** r_media);
- /*
+ /*
* Adds a sdes attribute to the given media section.
*
* @param media The media to add the srtp attribute to
- */
- void sdp_add_sdes_attribute(std::vector<std::string>& crypto);
+ */
+ void sdp_add_sdes_attribute(std::vector<std::string>& crypto);
/*
* Adds a zrtp-hash attribute to
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 4d068c4862..c38a29e979 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -820,30 +820,20 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
bool
SIPVoIPLink::answer (const CallID& id)
{
- SIPCall *call;
pj_status_t status;
pjsip_tx_data *tdata;
- Sdp *local_sdp;
pjsip_inv_session *inv_session;
- _debug ("SIPVoIPLink::answer: start answering ");
+ _debug ("UserAgent: Answering call %s", id.c_str());
- call = getSIPCall (id);
+ SIPCall *call = getSIPCall (id);
if (call==0) {
- _debug ("! SIP Failure: SIPCall doesn't exists");
+ _debug ("UserAgent: SIPCall %s doesn't exists while answering", id.c_str());
return false;
}
- local_sdp = call->getLocalSDP();
-
- /*
- try {
- call->getAudioRtp()->initAudioRtpSession (call);
- } catch (...) {
- _debug ("Failed to create rtp thread from answer");
- }
- */
+ Sdp *local_sdp = call->getLocalSDP();
inv_session = call->getInvSession();
@@ -3683,7 +3673,6 @@ mod_on_rx_request (pjsip_rx_data *rdata)
return false;
}
-
/* Create the local dialog (UAS) */
status = pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog);
if (status != PJ_SUCCESS) {
diff --git a/sflphone-common/src/sip/sipvoiplink.h b/sflphone-common/src/sip/sipvoiplink.h
index 8a3e45acd7..119c83c3dc 100644
--- a/sflphone-common/src/sip/sipvoiplink.h
+++ b/sflphone-common/src/sip/sipvoiplink.h
@@ -422,7 +422,7 @@ class SIPVoIPLink : public VoIPLink
* General Sip transport creation method according to the
* transport type specified in account settings
* @param id The account id for which a transport must
- * be created.
+ * be created.
*/
bool createSipTransport(AccountID id);
@@ -436,28 +436,28 @@ class SIPVoIPLink : public VoIPLink
*/
bool addTransportToMap(std::string key, pjsip_transport* transport);
- /**
+ /**
* Create SIP UDP transport from account's setting
* @param id The account id for which a transport must
- * be created.
+ * be created.
* @return pj_status_t PJ_SUCCESS on success
*/
int createUdpTransport (AccountID = "");
- /**
- * Create a TLS transport from the default TLS listener from
- * @param id The account id for which a transport must
- * be created.
- * @return pj_status_t PJ_SUCCESS on success
- */
- pj_status_t createTlsTransport(const AccountID& id, std::string remoteAddr);
+ /**
+ * Create a TLS transport from the default TLS listener from
+ * @param id The account id for which a transport must
+ * be created.
+ * @return pj_status_t PJ_SUCCESS on success
+ */
+ pj_status_t createTlsTransport(const AccountID& id, std::string remoteAddr);
/**
- * Create a UDP transport using stun server to resove public address
- * @param id The account id for which a transport must
- * be created.
- * @return pj_status_t PJ_SUCCESS on success
- */
+ * Create a UDP transport using stun server to resove public address
+ * @param id The account id for which a transport must
+ * be created.
+ * @return pj_status_t PJ_SUCCESS on success
+ */
pj_status_t createAlternateUdpTransport (AccountID id);
@@ -474,11 +474,11 @@ class SIPVoIPLink : public VoIPLink
EventThread* _evThread;
ost::Mutex _mutexSIP;
- /* Number of SIP accounts connected to the link */
- int _clients;
+ /* Number of SIP accounts connected to the link */
+ int _clients;
- /*
- * Get the correct address to use (ie advertised) from
+ /*
+ * Get the correct address to use (ie advertised) from
* a uri. The corresponding transport that should be used
* with that uri will be discovered.
*
--
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