diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index a081f88dce9d67d19859a772607e2f3c7a9dd573..e5fdf99b8c65a769e6710a80c309a865b0281d66 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -110,8 +110,7 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
_rtpSession = new AudioSrtpSession (&Manager::instance(), ca);
_rtpSessionType = Sdes;
- // ca->getLocalSDP()->set_srtp_master_key (static_cast<AudioSrtpSession *> (_rtpSession)->getMasterKey());
-
+ ca->getLocalSDP()->set_srtp_crypto(static_cast<AudioSrtpSession *> (_rtpSession)->getCryptoSdpInfo());
break;
default:
diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
index 67a07d1311ab1de4f180c2de3fc62615cf5e9789..d6fd2b273ca9861f58ff08b62022a3c013cea883 100644
--- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
@@ -36,12 +36,6 @@ static uint8 ms[] = { 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17,
0x18, 0x19, 0x1a, 0x1b, 0x1c, 0x1d };
-// static std::string crypto_suite = "AES_CM_128_HMAC_SHA1_32";
-// static std::string application = "srtp";
-// static std::string srtp_key = "inline:16/14/NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj/2^20/1:32";
-
-
-
namespace sfl
{
@@ -61,13 +55,26 @@ AudioSrtpSession::AudioSrtpSession (ManagerImpl * manager, SIPCall * sipcall) :
setOutQueueCryptoContext(outputCryptoCtx);
}
- /*
-std::string AudioSrtpSession::getCryptoInfo() {
+
+std::string AudioSrtpSession::getCryptoSdpInfo() {
+ std::string tag = "1";
+ std::string crypto_suite = "AES_CM_128_HMAC_SHA1_32";
+ std::string application = "srtp";
+ std::string srtp_key = "inline:16/14/NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj/2^20/1:32";
- return ;
+ std::string crypto = tag;
+
+ crypto.append(" ");
+ crypto.append(crypto_suite);
+ crypto.append(" ");
+ crypto.append(application);
+ crypto.append(" ");
+ crypto.append(srtp_key);
+
+ return crypto;
}
- */
+
void AudioSrtpSession::initializeMasterKey(void)
{
diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
index 89bc2450ed7b565c042ea676efee55e97cb71790..34adb9d505652afa1df9b7a9f96074ef0008bd46 100644
--- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
@@ -42,9 +42,7 @@ namespace sfl {
AudioSrtpSession(ManagerImpl * manager, SIPCall * sipcall);
- // std::string getCryptoInfo(void);
-
- uint8* getMasterKey(void){ return _masterKey; }
+ std::string getCryptoSdpInfo(void);
private:
diff --git a/sflphone-common/src/sip/sdp.cpp b/sflphone-common/src/sip/sdp.cpp
index 524d6f9fc1f8fd42b3915e57a62b1b9bcbc7f017..e069369c6188281c724aa0c31ca2df9c841a0eec 100644
--- a/sflphone-common/src/sip/sdp.cpp
+++ b/sflphone-common/src/sip/sdp.cpp
@@ -158,10 +158,9 @@ int Sdp::create_local_offer ()
//sdp_addAttributes( _pool );
sdp_add_media_description();
- // if(!_srtp_master_key.empty()) {
-
- sdp_add_sdes_attribute();
- // }
+ if(!_srtp_crypto.empty()) {
+ sdp_add_sdes_attribute(_srtp_crypto);
+ }
//toString ();
@@ -368,7 +367,7 @@ void Sdp::sdp_add_media_description()
}
-void Sdp::sdp_add_sdes_attribute ()
+void Sdp::sdp_add_sdes_attribute (std::string crypto)
{
char tempbuf[256];
@@ -382,12 +381,17 @@ void Sdp::sdp_add_sdes_attribute ()
attribute->name = pj_strdup3(_pool, "crypto");
+ /*
int len = pj_ansi_snprintf(tempbuf, sizeof(tempbuf),
"%.*s %.*s %.*s",
(int)tag.size(), tag.c_str(),
(int)crypto_suite.size(), crypto_suite.c_str(),
(int)key.size(), key.c_str());
-
+ */
+
+ int len = pj_ansi_snprintf(tempbuf, sizeof(tempbuf),
+ "%.*s",(int)crypto.size(), crypto.c_str());
+
attribute->value.slen = len;
attribute->value.ptr = (char*) pj_pool_alloc (_pool, attribute->value.slen+1);
pj_memcpy (attribute->value.ptr, tempbuf, attribute->value.slen+1);
diff --git a/sflphone-common/src/sip/sdp.h b/sflphone-common/src/sip/sdp.h
index 9f5fc9ea44902c909e99e1f13b47858590055bf5..72ef3f434cf310ae442f36d81f8ce4ecec6d8135 100644
--- a/sflphone-common/src/sip/sdp.h
+++ b/sflphone-common/src/sip/sdp.h
@@ -105,7 +105,7 @@ class Sdp {
/* Set the srtp _master_key
* @param mk The Master Key of a srtp session.
*/
- inline void set_srtp_master_key(const std::string& mk) { _srtp_master_key = mk; }
+ inline void set_srtp_crypto(const std::string& mk) { _srtp_crypto = mk; }
/*
* On building an invite outside a dialog, build the local offer and create the
@@ -253,7 +253,8 @@ class Sdp {
std::string _zrtp_hello_hash;
- std::string _srtp_master_key;
+ /** "a=crypto" sdes attribute obtained from AudioSrtpSession */
+ std::string _srtp_crypto;
Sdp(const Sdp&); //No Copy Constructor
Sdp& operator=(const Sdp&); //No Assignment Operator
@@ -344,14 +345,13 @@ class Sdp {
void get_remote_sdp_media_from_offer (const pjmedia_sdp_session* r_sdp, pjmedia_sdp_media** r_media);
void get_remote_sdp_crypto_from_offer (const pjmedia_sdp_session* remote_sdp, pjmedia_sdp_attr** r_crypto);
-
/*
* Adds a sdes attribute to the given media section.
*
* @param media The media to add the srtp attribute to
*/
- void sdp_add_sdes_attribute();
+ void sdp_add_sdes_attribute(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 bed479fc1654788e6ff069287cd6f05e091a20eb..4c0431382711c131c4b9e0e672c1eca25e277045 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -1559,20 +1559,22 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
setCallAudioLocal (call, localAddress);
_debug ("toUri received in new_ip_to_ip call %s", to.c_str());
-
std::string toUri = account->getToUri (to);
call->setPeerNumber (toUri);
_debug ("toUri in new_ip_to_ip call %s", toUri.c_str());
- // Building the local SDP offer
- call->getLocalSDP()->set_ip_address (addrSdp);
- call->getLocalSDP()->create_initial_offer();
- try {
+ // Audio Rtp Session must be initialized before creating initial offer in SDP session
+ // since SDES require crypto attribute.
+ try {
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("! SIP Failure: Unable to create RTP Session in SIPVoIPLink::new_ip_to_ip_call (%s:%d)", __FILE__, __LINE__);
}
+ // Building the local SDP offer
+ call->getLocalSDP()->set_ip_address (addrSdp);
+ call->getLocalSDP()->create_initial_offer();
+
// If no account already set, use the default one created at pjsip initialization
if (account->getAccountTransport() == NULL) {
_debug ("No transport for this account, using the default one");