[asterisk-scf-commits] asterisk-scf/integration/media_rtp_pjmedia.git branch "nat-support" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Wed Jul 6 09:27:26 CDT 2011
branch "nat-support" has been updated
via 3985edfb47c8d88392ed6e39e7f82bc0862e7185 (commit)
via 0492b17c7c9461d4df682ad6f742680723411186 (commit)
from cfbf9175d04aa73f0d379a3342f1c691989bf182 (commit)
Summary of changes:
src/ICETransport.cpp | 140 +++++++++++++++++++++++++++++++++++++++++----
src/PJMediaEnvironment.h | 6 +-
src/PJMediaTransport.cpp | 8 +-
src/SRTPConfiguration.cpp | 2 +-
src/SessionAdapter.h | 13 ++--
src/UDPTransport.cpp | 44 ++++----------
src/UDPTransport.h | 5 ++
7 files changed, 161 insertions(+), 57 deletions(-)
- Log -----------------------------------------------------------------
commit 3985edfb47c8d88392ed6e39e7f82bc0862e7185
Author: Brent Eagles <beagles at digium.com>
Date: Mon Jul 4 14:25:22 2011 -0230
Incorporated review feedback.
diff --git a/src/PJMediaEnvironment.h b/src/PJMediaEnvironment.h
index b9be64a..dc4d39d 100644
--- a/src/PJMediaEnvironment.h
+++ b/src/PJMediaEnvironment.h
@@ -84,7 +84,7 @@ public:
}
/**
- * Get SRTP relatd configuratino object.
+ * Get SRTP related configuration object.
*/
SRTPConfigurationPtr srtpConfig() const
{
@@ -93,7 +93,7 @@ public:
/**
* Get our library instance's main pool factory. As this function does not return a reference counted pointer, its
- * value should not be cached at all. It's validity is directly related to the lifetime of the PJMediaEnvironment
+ * value should not be cached at all. Its validity is directly related to the lifetime of the PJMediaEnvironment
* object.
*/
pj_pool_factory* poolFactory() const
@@ -103,7 +103,7 @@ public:
/**
* Get the main memory pool. As this function does not return a reference counted pointer, its value should
- * not be cached at all. It's validity is directly related to the lifetime of the PJMediaEnvironment object.
+ * not be cached at all. Its validity is directly related to the lifetime of the PJMediaEnvironment object.
*/
pj_pool_t* memoryPool() const
{
diff --git a/src/PJMediaTransport.cpp b/src/PJMediaTransport.cpp
index f218831..c646175 100644
--- a/src/PJMediaTransport.cpp
+++ b/src/PJMediaTransport.cpp
@@ -52,13 +52,13 @@ AddressPtr PJMediaTransport::remoteAddress()
pjmedia_transport_info_init(&info);
pjmedia_transport_get_info(mTransport, &info);
- if (info.src_rtp_name.addr.sa_family != PJ_AF_INET &&
- info.src_rtp_name.addr.sa_family != PJ_AF_INET6)
+ if (info.src_rtp_name.addr.sa_family != pj_AF_INET() &&
+ info.src_rtp_name.addr.sa_family != pj_AF_INET6())
{
//
// If we don't have the remote address yet (we know because the default
- // intialization for the above is neither PF_AF_INET or PF_AF_INET6) then
- // return 0
+ // intialization for the above is neither pj_AF_INET() or
+ // pj_AF_INET6()) then return 0
//
return AddressPtr();
}
diff --git a/src/SRTPConfiguration.cpp b/src/SRTPConfiguration.cpp
index e2acc49..9ae53ae 100644
--- a/src/SRTPConfiguration.cpp
+++ b/src/SRTPConfiguration.cpp
@@ -24,7 +24,7 @@ SRTPConfigurationPtr AsteriskSCF::PJMediaRTP::SRTPConfiguration::create(const Ic
const string& propertyPrefix)
{
string prefix(propertyPrefix);
- if (!prefix.empty() && *(--prefix.end()) != '.')
+ if (!prefix.empty() && prefix[prefix.length()-1] != '.')
{
prefix += '.';
}
diff --git a/src/SessionAdapter.h b/src/SessionAdapter.h
index 0c550d1..d164bca 100644
--- a/src/SessionAdapter.h
+++ b/src/SessionAdapter.h
@@ -33,12 +33,13 @@ namespace PJMediaRTP
class SessionAdapter;
typedef boost::shared_ptr<SessionAdapter> SessionAdapterPtr;
-/**
- * By formalizing the interface between the session servant and the source and sink servants, we can
- * remove the physical compile time interdepencies. For the moment, this means that there will be
- * an indirect circular reference between the three entities. This is fine for the moment because
- * they are actually quite tightly coupled implementation-wise. e.g. setRemoteDetails propogates to the
- * source servant.
+/**
+ * By formalizing the interface between the session servant and the source and
+ * sink servants, we can remove the physical compile time interdepencies. For
+ * the moment, this means that there will be an indirect circular reference
+ * between the three entities. This is fine for the moment because they are
+ * actually quite tightly coupled implementation-wise. e.g. setRemoteDetails
+ * propagates to the source servant.
**/
class SessionAdapter
{
diff --git a/src/UDPTransport.cpp b/src/UDPTransport.cpp
index 18d9c17..4294c29 100644
--- a/src/UDPTransport.cpp
+++ b/src/UDPTransport.cpp
@@ -31,7 +31,19 @@ UDPTransportPtr AsteriskSCF::PJMediaRTP::UDPTransport::create(const PJMediaEndpo
{
const Ice::Int minPort = configObject->getStartPort();
const Ice::Int maxPort = configObject->getEndPort();
+ return createImpl(ep, configObject, minPort, maxPort, expectIPv6);
+}
+UDPTransportPtr AsteriskSCF::PJMediaRTP::UDPTransport::create(const PJMediaEndpointPtr& ep,
+ const RTPConfigurationPtr& configObject, unsigned port, bool expectIPv6)
+{
+ return createImpl(ep, configObject, port, port + 1, expectIPv6);
+}
+
+UDPTransportPtr AsteriskSCF::PJMediaRTP::UDPTransport::createImpl(const PJMediaEndpointPtr& ep,
+ const RTPConfigurationPtr& configObject, unsigned minPort, unsigned maxPort,
+ bool expectIPv6)
+{
int addressFamily = pj_AF_INET();
string bindingAddress;
if (expectIPv6)
@@ -68,38 +80,6 @@ UDPTransportPtr AsteriskSCF::PJMediaRTP::UDPTransport::create(const PJMediaEndpo
throw InternalInitializationException("Unable to initialize UDP media transport");
}
-UDPTransportPtr AsteriskSCF::PJMediaRTP::UDPTransport::create(const PJMediaEndpointPtr& ep,
- const RTPConfigurationPtr& configObject, unsigned port, bool expectIPv6)
-{
- int addressFamily = pj_AF_INET();
- string bindingAddress;
- if (expectIPv6)
- {
- addressFamily = pj_AF_INET6();
- bindingAddress = configObject->getBindIPv6Address();
- }
- else
- {
- bindingAddress = configObject->getBindIPv4Address();
- }
- pj_str_t pjBindAddress;
- pj_strset(&pjBindAddress, (char*)bindingAddress.c_str(), bindingAddress.size());
- pj_str_t* bindAddressPtr = 0;
- if (!bindingAddress.empty())
- {
- bindAddressPtr = &pjBindAddress;
- }
-
- pjmedia_transport* transport;
- pj_status_t result = pjmedia_transport_udp_create3(ep->endpoint(), addressFamily, "RTP", bindAddressPtr,
- port, 0, &transport);
- if (success(result))
- {
- return UDPTransportPtr(new UDPTransport(transport));
- }
- throw InternalInitializationException("Unable to initialize UDP media transport");
-}
-
UDPTransport::UDPTransport(pjmedia_transport* t) :
PJMediaTransport(t)
{
diff --git a/src/UDPTransport.h b/src/UDPTransport.h
index 6e50e93..57283af 100644
--- a/src/UDPTransport.h
+++ b/src/UDPTransport.h
@@ -56,6 +56,11 @@ public:
bool expectIPv6);
private:
+
+ static UDPTransportPtr createImpl(const PJMediaEndpointPtr& ep,
+ const RTPConfigurationPtr& configObject,
+ unsigned minPort, unsigned maxPort, bool expectIPv6);
+
UDPTransport(pjmedia_transport* t);
};
commit 0492b17c7c9461d4df682ad6f742680723411186
Author: Brent Eagles <beagles at digium.com>
Date: Tue Jul 5 11:58:00 2011 -0230
Rework offer processing lost in merge.
diff --git a/src/ICETransport.cpp b/src/ICETransport.cpp
index 07d9ed3..6cbfc00 100644
--- a/src/ICETransport.cpp
+++ b/src/ICETransport.cpp
@@ -57,7 +57,8 @@ public:
mId(id),
mShuttingDown(false),
mNATType(AsteriskSCF::System::NAT::V1::Unknown),
- mRole(UndefinedRole)
+ mRole(UndefinedRole),
+ mTransport(0)
{
}
@@ -80,13 +81,131 @@ public:
return mRole;
}
- CandidatePtr negotiate(const CandidateSeq&,
+ CandidatePtr negotiate(const string& hostname, Ice::Int port, const CandidateSeq& candidates,
const Ice::Current&)
{
//
- // TODO: implement.
+ // So how this works is we create a remote SDP and call pjmedia_transport_start() easy peasy. (Same deal
//
- return 0;
+ pjmedia_sdp_session* remoteSDPSession =
+ static_cast<pjmedia_sdp_session*>(pj_pool_zalloc(mEnv->memoryPool(), sizeof(pjmedia_sdp_session)));
+
+
+ //
+ // TODO: I think the ICE transport ignores a lot of this stuff, but I'm going to add it for the time
+ // being anyways.
+ //
+
+ //
+ // Missing details, user, id, version, net type?
+ //
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->name, "ASCFMEDIA");
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->origin.user, "");
+ AddressPtr remoteHost(new Address(hostname, port));
+
+ remoteSDPSession->conn = static_cast<pjmedia_sdp_conn*>(pj_pool_zalloc(mEnv->memoryPool(), sizeof(pjmedia_sdp_conn)));
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->conn->net_type, "IN");
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->origin.net_type, "IN");
+
+ //
+ // TODO: Look at whether the members can point to the same memory without issues.
+ //
+ if (remoteHost->isIPV6())
+ {
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->origin.addr_type, "IP6");
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->conn->addr_type, "IP6");
+ }
+ else
+ {
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->origin.addr_type, "IP4");
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->conn->addr_type, "IP4");
+ }
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->origin.addr, remoteHost->address().c_str());
+ pj_strdup2(mEnv->memoryPool(), &remoteSDPSession->conn->addr, remoteHost->address().c_str());
+ remoteSDPSession->attr_count = 0;
+
+ //
+ // Cut n' paste from current SIP session manager... icky. TODO: sift through and see what of this
+ // can be discarded for this purpose.
+ //
+ remoteSDPSession->media_count = 1;
+ pjmedia_sdp_media* media =
+ static_cast<pjmedia_sdp_media*>(pj_pool_zalloc(mEnv->memoryPool(), sizeof(pjmedia_sdp_media)));
+ remoteSDPSession->media[0] = media;
+ pj_strdup2(mEnv->memoryPool(), &media->desc.media, "audio");
+ media->desc.port = (pj_uint16_t) port; // XXX --- this is not going to be correct here.. we don't actually have this!
+ media->desc.port_count = 1;
+ pj_strdup2(mEnv->memoryPool(), &media->desc.transport, "RTP/AVP");
+
+ // Populate the stream with codec details
+ remoteSDPSession->media[0]->desc.fmt_count = 1;
+ remoteSDPSession->media[0]->attr_count = 0;
+
+ // TODO: We should iterate over the formats to produce this instead of hardcoding
+ pjmedia_sdp_rtpmap rtpmap;
+ pjmedia_sdp_attr *attr;
+
+ // This is hardcoded value for ULAW for now
+ pj_strdup2(mEnv->memoryPool(), &media->desc.fmt[0], "0");
+ rtpmap.pt = media->desc.fmt[0];
+ rtpmap.clock_rate = 8000;
+ pj_strdup2(mEnv->memoryPool(), &rtpmap.enc_name, "PCMU");
+ rtpmap.param.slen = 0;
+ pjmedia_sdp_rtpmap_to_attr(mEnv->memoryPool(), &rtpmap, &attr);
+ remoteSDPSession->media[0]->attr[remoteSDPSession->media[0]->attr_count++] = attr;
+
+ // Might as well add sendrecv
+ attr = static_cast<pjmedia_sdp_attr*>(pj_pool_zalloc(mEnv->memoryPool(), sizeof(pjmedia_sdp_attr)));
+ pj_strdup2(mEnv->memoryPool(), &attr->name, "sendrecv");
+ remoteSDPSession->media[0]->attr[remoteSDPSession->media[0]->attr_count++] = attr;
+
+ //
+ // I was concerned about the fact that for a given SIP session, there might be multiple media
+ // streams and multiple candidates. I'm not sure that its actually too much of an issue even
+ // if multiple media types are muxed on a single ICE negotiated flow, but there will need to be
+ // some redesign to pull in the multiple media streams associated with the session. For the moment
+ // we will operation under the premise that we are dealing with a single media stream.
+ // TODO: the SIP session gateway contains similar code, but from the offer perspective. This stuff
+ // should be refactored into a pjproject utility library.
+ //
+ //
+ pjmedia_sdp_media* currentMedia = remoteSDPSession->media[0];
+ for (CandidateSeq::const_iterator i = candidates.begin(); i != candidates.end(); ++i)
+ {
+ CandidatePtr candidate = *i;
+ ostringstream os;
+ os << "candidate:" << candidate->foundation << ' ' << candidate->componentId << " UDP " <<
+ candidate->priority << ' ' << candidate->mappedAddress << ' ' << candidate->mappedPort << " typ ";
+ string hostType;
+ switch (candidate->type)
+ {
+ case Host:
+ hostType = "host";
+ break;
+ case ServerReflexive:
+ hostType = "srflx";
+ break;
+ case PeerReflexive:
+ hostType = "prflx";
+ break;
+ case Relayed:
+ hostType = "relay";
+ break;
+ }
+ os << hostType;
+ if (candidate->type != Host)
+ {
+ os << " raddr " << candidate->baseAddress << " rport " << candidate->basePort;
+ }
+ string t = os.str();
+ pj_str_t candidateStr = pj_str(const_cast<char*>(t.c_str()));
+ pjmedia_sdp_attr* newAttribute = pjmedia_sdp_attr_create(mEnv->memoryPool(),
+ "candidate", &candidateStr);
+ pjmedia_sdp_attr_add(¤tMedia->attr_count, currentMedia->attr, newAttribute);
+ }
+ pjmedia_sdp_session localSession;
+ pjmedia_transport_encode_sdp(mTransport, mEnv->memoryPool(), &localSession, 0, 0);
+ pjmedia_transport_media_start(mTransport, mEnv->memoryPool(), &localSession, remoteSDPSession, 0);
}
CandidateSeq getCandidates(const Ice::Current&)
@@ -104,6 +223,7 @@ public:
//
return;
}
+ mTransport = transport;
pjmedia_transport_info info;
pjmedia_transport_info_init(&info);
@@ -141,15 +261,12 @@ public:
// candidate structures, so what we have to do is get the SDP from
// the transport and convert what we find there to our Ice structures.
//
- pjmedia_sdp_session* sdpSession;
- //
- // TODO: We are setting the transport size at 1, but I'm not sure that will always be the case.
- //
- pjmedia_endpt_create_sdp(mEndpoint->endpoint(), mEnv->memoryPool(), 1, &info.sock_info, &sdpSession);
- for (size_t i = 0; i < sdpSession->media_count; ++i)
+ pjmedia_sdp_session sdpSession;
+ pjmedia_transport_encode_sdp(mTransport, mEnv->memoryPool(), &sdpSession, 0, 0);
+ for (size_t i = 0; i < sdpSession.media_count; ++i)
{
const string candidateName("candidate");
- pjmedia_sdp_media* media = sdpSession->media[i];
+ pjmedia_sdp_media* media = sdpSession.media[i];
for (size_t j = 0; j < media->attr_count; ++j)
{
pjmedia_sdp_attr* attr = media->attr[j];
@@ -329,6 +446,7 @@ private:
CandidateSeq mCandidates;
PJMediaEnvironmentPtr mEnv;
PJMediaEndpointPtr mEndpoint;
+ pjmedia_transport* mTransport;
void stateCheck()
{
-----------------------------------------------------------------------
--
asterisk-scf/integration/media_rtp_pjmedia.git
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