From 39c084ffb7cff4e33256a0c843461b4300c5e816 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Wed, 20 Apr 2011 18:33:34 -0400
Subject: [PATCH] [#5571] Move attribute handling for onhold/offhold actions in
SDP session
---
.../src/audio/audiortp/AudioRtpFactory.cpp | 4 +-
sflphone-common/src/sip/sdp.cpp | 15 ++
sflphone-common/src/sip/sdp.h | 44 +++--
sflphone-common/src/sip/sipvoiplink.cpp | 175 +++++++++---------
sflphone-common/src/sip/sipvoiplink.h | 8 +-
5 files changed, 139 insertions(+), 107 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index cc52b96c63..8c8d54d1db 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -127,7 +127,7 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
if (_helloHashEnabled) {
// TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
// not even available at that point.
- ca->getLocalSDP()->set_zrtp_hash (static_cast<AudioZrtpSession *> (_rtpSession)->getHelloHash());
+ ca->getLocalSDP()->setZrtpHash (static_cast<AudioZrtpSession *> (_rtpSession)->getHelloHash());
_debug ("AudioRtpFactory: Zrtp hello hash fed to SDP");
}
@@ -320,7 +320,7 @@ void sfl::AudioRtpFactory::initLocalCryptoInfo (SIPCall * ca)
if (_rtpSession && _rtpSessionType && (_rtpSessionType == Sdes)) {
static_cast<AudioSrtpSession *> (_rtpSession)->initLocalCryptoInfo ();
- ca->getLocalSDP()->set_srtp_crypto (static_cast<AudioSrtpSession *> (_rtpSession)->getLocalCryptoInfo());
+ ca->getLocalSDP()->setLocalSdpCrypto (static_cast<AudioSrtpSession *> (_rtpSession)->getLocalCryptoInfo());
}
}
diff --git a/sflphone-common/src/sip/sdp.cpp b/sflphone-common/src/sip/sdp.cpp
index b921c47e2d..5a5d62532b 100644
--- a/sflphone-common/src/sip/sdp.cpp
+++ b/sflphone-common/src/sip/sdp.cpp
@@ -717,6 +717,21 @@ void Sdp::setPortToAllMedia (int port)
}
}
+void Sdp::addAttributeToLocalAudioMedia(std::string attr)
+{
+ pjmedia_sdp_attr *attribute;
+
+ attribute = pjmedia_sdp_attr_create (memPool, attr.c_str(), NULL);
+
+ pjmedia_sdp_media_add_attr (getLocalSdpSession()->media[0], attribute);
+}
+
+void Sdp::removeAttributeFromLocalAudioMedia(std::string attr)
+{
+ pjmedia_sdp_media_remove_all_attr (getLocalSdpSession()->media[0], attr.c_str());
+
+}
+
std::string Sdp::convertIntToString (int value)
{
std::ostringstream result;
diff --git a/sflphone-common/src/sip/sdp.h b/sflphone-common/src/sip/sdp.h
index e44fcf5fa3..73ccecb185 100644
--- a/sflphone-common/src/sip/sdp.h
+++ b/sflphone-common/src/sip/sdp.h
@@ -282,30 +282,52 @@ class Sdp
return remoteAudioPort;
}
- std::vector<sdpMedia*> getSessionMediaList (void) {
+ /**
+ * Get media list for this session
+ */
+ SdpMediaList getSessionMediaList (void) {
return sessionAudioMedia;
}
+ /**
+ *
+ */
+ void addAttributeToLocalAudioMedia(std::string);
+
+ /**
+ *
+ */
+ void removeAttributeFromLocalAudioMedia(std::string);
+
+
+ /**
+ * Get SRTP master key
+ * @param remote sdp session
+ * @param crypto offer
+ */
void getRemoteSdpCryptoFromOffer (const pjmedia_sdp_session* remote_sdp, CryptoOffer& crypto_offer);
+ /**
+ * Set the SRTP master_key
+ * @param mk The Master Key of a srtp session.
+ */
+ inline void setLocalSdpCrypto (const std::vector<std::string> lc) {
+ srtpCrypto = lc;
+ }
- /* Set the zrtp hash that was previously calculated from the hello message in the zrtp layer.
+ /**
+ * Set the zrtp hash that was previously calculated from the hello message in the zrtp layer.
* This hash value is unique at the media level. Therefore, if video support is added, one would
* have to set the correct zrtp-hash value in the corresponding media section.
* @param hash The hello hash of a rtp session. (Only audio at the moment)
*/
- inline void set_zrtp_hash (const std::string& hash) {
+ inline void setZrtpHash (const std::string& hash) {
zrtpHelloHash = hash;
- _debug ("Zrtp hash set with %s\n", hash.c_str());
- }
-
- /* Set the srtp _master_key
- * @param mk The Master Key of a srtp session.
- */
- inline void set_srtp_crypto (const std::vector<std::string> lc) {
- srtpCrypto = lc;
}
+ /**
+ * Print internal state info
+ */
void toString (void);
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 87a501d4b8..9302e9c9ef 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -897,35 +897,109 @@ SIPVoIPLink::cancel (const CallID& id)
return true;
}
+
bool
SIPVoIPLink::onhold (const CallID& id)
{
-
+ Sdp *sdpSession;
pj_status_t status;
SIPCall* call;
call = getSIPCall (id);
- if (call==0) {
- _debug ("! SIP Error: call doesn't exist");
+ if (call == NULL) {
+ _debug ("UserAgent: Error: call doesn't exist in onhold action");
return false;
}
-
// Stop sound
call->setAudioStart (false);
-
call->setState (Call::Hold);
+ call->getAudioRtp()->stop();
- _debug ("* SIP Info: Stopping AudioRTP for onhold action");
+ _debug ("UserAgent: Stopping RTP session for on hold action");
- call->getAudioRtp()->stop();
+ if ( (sdpSession = call->getLocalSDP()) == NULL) {
+ _debug ("UserAgent: Error: Unable to find local sdp");
+ return false;
+ }
+
+ sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
+ sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
+
+ sdpSession->addAttributeToLocalAudioMedia("sendonly");
+
+ // Create re-INVITE with new offer
+ status = SIPSessionReinvite (call);
+
+ if (status != PJ_SUCCESS) {
+ return false;
+ }
+
+ return true;
+}
+
+bool
+SIPVoIPLink::offhold (const CallID& id)
+{
+ Sdp *sdpSession;
+ pj_status_t status;
+ SIPCall *call;
+
+ _debug ("UserAgent: retrive call from hold status");
+
+ call = getSIPCall (id);
+
+ if (call==0) {
+ _debug ("! SIP Error: Call doesn't exist");
+ return false;
+ }
+
+ if ( (sdpSession = call->getLocalSDP()) == NULL) {
+ _debug ("UserAgent: Error: Unable to find local sdp");
+ return false;
+ }
+
+ // Retreive previously selected codec
+ sfl::Codec *sessionMedia = sdpSession->getSessionMedia();
+
+ if (!sessionMedia) {
+ return false;
+ }
+
+ // Get PayloadType for this codec
+ AudioCodecType pl = (AudioCodecType) sessionMedia->getPayloadType();
+
+ _debug ("UserAgent: Payload from session media %d", pl);
+
+ try {
+ // Create a new instance for this codec
+ sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (pl);
+
+ if (audiocodec == NULL)
+ _error ("UserAgent: No audiocodec found");
+
+ call->getAudioRtp()->initAudioRtpConfig (call);
+ call->getAudioRtp()->initAudioRtpSession (call);
+ call->getAudioRtp()->start (static_cast<AudioCodec *>(audiocodec));
+
+ } catch (...) {
+ _debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
+ }
+
+ sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
+ sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
+
+ sdpSession->addAttributeToLocalAudioMedia("sendrecv");
/* Create re-INVITE with new offer */
- status = SIPInvSessionReinvite (call, "sendonly");
+ status = SIPSessionReinvite (call);
- if (status != PJ_SUCCESS)
+ if (status != PJ_SUCCESS) {
return false;
+ }
+
+ call->setState (Call::Active);
return true;
}
@@ -968,14 +1042,12 @@ SIPVoIPLink::sendTextMessage (sfl::InstantMessaging *module, const std::string&
return status;
}
-int SIPVoIPLink::SIPInvSessionReinvite (SIPCall *call, std::string direction)
+int SIPVoIPLink::SIPSessionReinvite (SIPCall *call)
{
pj_status_t status;
pjsip_tx_data *tdata;
pjmedia_sdp_session *local_sdp;
- pjmedia_sdp_attr *attr;
- pj_pool_t *call_memory_pool;
if (call == NULL) {
_error ("UserAgent: Error: Call is NULL in session reinvite");
@@ -987,32 +1059,8 @@ int SIPVoIPLink::SIPInvSessionReinvite (SIPCall *call, std::string direction)
return !PJ_SUCCESS;
}
- if ( (call_memory_pool = call->getLocalSDP()->getMemoryPool()) == NULL) {
- _debug ("UserAgent: Error: Unable to find call memory pool");
- return !PJ_SUCCESS;
- }
-
- // Reinvite only if connected
- // Build the local SDP offer
- // TODO Restore Re-Invite
- // status = call->getLocalSDP()->create_initial_offer();
-
- // if (status != PJ_SUCCESS)
- // return 1; // !PJ_SUCCESS
-
-
- pjmedia_sdp_media_remove_all_attr (local_sdp->media[0], "sendrecv");
- pjmedia_sdp_media_remove_all_attr (local_sdp->media[0], "sendonly");
-
- attr = pjmedia_sdp_attr_create (call_memory_pool, direction.c_str(), NULL);
-
- pjmedia_sdp_media_add_attr (local_sdp->media[0], attr);
-
- // pjmedia_sdp_neg_modify_local_offer (_pool, call->getLocalSDP()->_negociator, local_sdp);
-
// Build the reinvite request
- status = pjsip_inv_reinvite (call->getInvSession(), NULL,
- local_sdp, &tdata);
+ status = pjsip_inv_reinvite (call->getInvSession(), NULL, local_sdp, &tdata);
if (status != PJ_SUCCESS)
return 1; // !PJ_SUCCESS
@@ -1026,59 +1074,6 @@ int SIPVoIPLink::SIPInvSessionReinvite (SIPCall *call, std::string direction)
return PJ_SUCCESS;
}
-
-bool
-SIPVoIPLink::offhold (const CallID& id)
-{
- SIPCall *call;
- pj_status_t status;
-
- _debug ("UserAgent: retrive call from hold status");
-
- call = getSIPCall (id);
-
- if (call==0) {
- _debug ("! SIP Error: Call doesn't exist");
- return false;
- }
-
- // Retreive previously selected codec
- sfl::Codec *sessionMedia = call->getLocalSDP()->getSessionMedia();
-
- if (!sessionMedia)
- return false;
-
- // Get PayloadType for this codec
- AudioCodecType pl = (AudioCodecType) sessionMedia->getPayloadType();
-
- _debug ("UserAgent: Payload from session media %d", pl);
-
- try {
- // Create a new instance for this codec
- sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (pl);
-
- if (audiocodec == NULL)
- _error ("UserAgent: No audiocodec found");
-
- call->getAudioRtp()->initAudioRtpConfig (call);
- call->getAudioRtp()->initAudioRtpSession (call);
- call->getAudioRtp()->start (static_cast<AudioCodec *>(audiocodec));
-
- } catch (...) {
- _debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
- }
-
- /* Create re-INVITE with new offer */
- status = SIPInvSessionReinvite (call, "sendrecv");
-
- if (status != PJ_SUCCESS)
- return false;
-
- call->setState (Call::Active);
-
- return true;
-}
-
bool
SIPVoIPLink::transfer (const CallID& id, const std::string& to)
{
diff --git a/sflphone-common/src/sip/sipvoiplink.h b/sflphone-common/src/sip/sipvoiplink.h
index 76fc825538..abcd552abb 100644
--- a/sflphone-common/src/sip/sipvoiplink.h
+++ b/sflphone-common/src/sip/sipvoiplink.h
@@ -300,18 +300,18 @@ class SIPVoIPLink : public VoIPLink
/**
* Handle a re-invite request by the remote peer.
* A re-invite is an invite request inside a dialog.
- * When receiving a re-invite, we close the current rtp session and create
- * a new one with the updated information
+ * When receiving a re-invite, we updated information
+ * concerning medias
* @param sip call
*/
void SIPHandleReinvite (SIPCall *call);
/**
* Send a reINVITE inside an active dialog to modify its state
+ * Local SDP session should be modified before calling this method
* @param sip call
- * @param the direction, "sendrecv" or "sendonly"
*/
- int SIPInvSessionReinvite (SIPCall *call, std::string direction="");
+ int SIPSessionReinvite (SIPCall *call);
pj_caching_pool *getMemoryPoolFactory();
--
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