Added optional message to verify reINVITE in media proxying mode.
git-svn-id: http://voip.null.ro/svn/yate@5915 acf43c95-373e-0410-b603-e72c3f656dc1
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paulc
parent
61b39a94e6
commit
e13b4b1a7b
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@ -247,6 +247,9 @@
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; update_target: bool: Update dialog target from Contact in reINVITE
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;update_target=disable
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; update_verify: bool: Use a message to verify if we should accept a reINVITE when proxying media
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;update_verify=disable
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; preventive_bye: bool: If possible send a BYE besides CANCEL for unanswered calls
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;preventive_bye=enable
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@ -1000,6 +1000,8 @@ private:
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void emitUpdate();
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bool emitPRACK(const SIPMessage* msg);
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bool startClientReInvite(NamedList& msg, bool rtpForward);
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bool reInviteForward(SIPTransaction* t, MimeSdpBody* sdp, int invite);
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bool reInviteProxy(SIPTransaction* t, MimeSdpBody* sdp, int invite);
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// Build the 'call.route' and NOTIFY messages needed by the transfer thread
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bool initTransfer(Message*& msg, SIPMessage*& sipNotify, const SIPMessage* sipRefer,
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const MimeHeaderLine* refHdr, const URI& uri, const MimeHeaderLine* replaces);
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@ -1212,6 +1214,7 @@ static bool s_gen_async = true;
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static bool s_multi_ringing = false;
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static bool s_refresh_nosdp = true;
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static bool s_update_target = false;
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static bool s_update_verify = false;
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static bool s_preventive_bye = true;
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static bool s_ignoreVia = true; // Ignore Via headers and send answer back to the source
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static bool s_sipt_isup = false; // Control the application/isup body processing
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@ -6822,10 +6825,6 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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if (!checkUser(t))
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return;
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DDebug(this,DebugAll,"YateSIPConnection::reInvite(%p) [%p]",t,this);
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m_dialog.adjustCSeq(t->initialMessage());
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// Change party
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if (t->getEngine()->autoChangeParty() && t->initialMessage() && !t->initialMessage()->isOutgoing())
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setPartyChanged(t->initialMessage()->getParty(),this);
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Lock mylock(driver());
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int invite = m_reInviting;
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if (m_tr || m_tr2 || (invite == ReinviteRequest) || (invite == ReinviteReceived)) {
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@ -6839,17 +6838,38 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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}
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m_reInviting = ReinviteReceived;
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mylock.drop();
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m_dialog.adjustCSeq(t->initialMessage());
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// Change party
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if (t->getEngine()->autoChangeParty() && t->initialMessage() && !t->initialMessage()->isOutgoing())
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setPartyChanged(t->initialMessage()->getParty(),this);
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// hack: use a while instead of if so we can return or break out of it
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MimeSdpBody* sdp = getSdpBody(t->initialMessage()->body);
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while (sdp) {
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// for pass-trough RTP we need support from our peer
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if (m_rtpForward) {
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if (sdp) {
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if (m_rtpForward ? reInviteForward(t,sdp,invite) : reInviteProxy(t,sdp,invite))
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return;
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}
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m_reInviting = invite;
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if (s_refresh_nosdp && !sdp) {
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// be permissive, accept session refresh with no SDP
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updateTarget(t->initialMessage());
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SIPMessage* m = new SIPMessage(t->initialMessage(),200);
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// if required provide our own media offer
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if (!m_rtpForward)
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m->setBody(createSDP());
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t->setResponse(m);
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m->deref();
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return;
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}
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t->setResponse(488);
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}
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bool YateSIPConnection::reInviteForward(SIPTransaction* t, MimeSdpBody* sdp, int invite)
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{
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String addr;
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String natAddr;
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ObjList* lst = plugin.parser().parse(sdp,addr,0,String::empty(),true);
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if (!lst)
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break;
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return false;
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// guess if the call comes from behind a NAT
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if (s_auto_nat && isNatBetween(addr,m_host)) {
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Debug(this,DebugInfo,"RTP NAT detected: private '%s' public '%s'",
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@ -6859,6 +6879,7 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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}
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Debug(this,DebugAll,"reINVITE RTP addr '%s'",addr.c_str());
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// for pass-trough RTP we need support from our peer
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Message msg("call.update");
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complete(msg);
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if (s_update_target) {
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@ -6897,7 +6918,7 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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TelEngine::destruct(lst);
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addSdpParams(msg,sdp);
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bool ok = Engine::dispatch(msg);
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Lock mylock2(driver());
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Lock mylock(driver());
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// if peer doesn't support updates fail the reINVITE
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if (!ok) {
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t->setResponse(msg.getIntValue(YSTRING("error"),dict_errors,488),msg.getValue(YSTRING("reason")));
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@ -6908,7 +6929,7 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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t->setResponse(491);
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// media is uncertain now so drop the call
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setReason("Internal Server Error",500);
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mylock2.drop();
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mylock.drop();
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hangup();
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}
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else {
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@ -6917,26 +6938,54 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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t->setUserData(this);
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m_tr2 = t;
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}
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return;
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}
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return true;
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}
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bool YateSIPConnection::reInviteProxy(SIPTransaction* t, MimeSdpBody* sdp, int invite)
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{
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// refuse request if we had no media at all before
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if (m_mediaStatus == MediaMissing)
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break;
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return false;
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String addr;
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String natAddr;
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ObjList* lst = plugin.parser().parse(sdp,addr);
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if (!lst)
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break;
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return false;
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// guess if the call comes from behind a NAT
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if (s_auto_nat && isNatBetween(addr,m_host)) {
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Debug(this,DebugInfo,"RTP NAT detected: private '%s' public '%s'",
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addr.c_str(),m_host.c_str());
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natAddr = addr;
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addr = m_host;
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}
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// TODO: check if we should accept the new media
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// many implementation don't handle well failure so we should drop
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bool audioChg = (getMedia(YSTRING("audio")) != 0);
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audioChg ^= ((*lst)[YSTRING("audio")] != 0);
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Message ver("call.update");
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if (s_update_verify) {
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complete(ver);
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ver.addParam("operation","verify");
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ver.addParam("method",t->initialMessage()->method);
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copySipHeaders(ver,*t->initialMessage());
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ver.addParam("rtp_addr",addr);
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if (natAddr)
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ver.addParam("rtp_nat_addr",natAddr);
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ver.addParam("audio_changed",String::boolText(audioChg));
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putMedia(ver,lst);
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addSdpParams(ver,sdp);
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if (!Engine::dispatch(ver) || (ver.retValue() == YSTRING("error")) || (ver.retValue() == "-")) {
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TelEngine::destruct(lst);
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SIPMessage* m = new SIPMessage(t->initialMessage(),
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ver.getIntValue(YSTRING("error"),dict_errors,488),ver.getValue(YSTRING("reason")));
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copySipHeaders(*m,ver);
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t->setResponse(m);
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m->deref();
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m_reInviting = invite;
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return true;
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}
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}
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if (m_rtpAddr != addr) {
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m_rtpAddr = addr;
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Debug(this,DebugAll,"New RTP addr '%s'",m_rtpAddr.c_str());
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@ -6945,7 +6994,6 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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setMedia(0);
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}
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setMedia(lst);
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audioChg ^= (getMedia(YSTRING("audio")) != 0);
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m_mediaStatus = MediaMissing;
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// let RTP guess again the local interface or use the enforced address
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@ -6957,10 +7005,13 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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bool contactChg = (uri != m_uri);
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SIPMessage* m = new SIPMessage(t->initialMessage(), 200);
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copySipHeaders(*m,ver);
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MimeSdpBody* sdpNew = createRtpSDP(true);
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m->setBody(sdpNew);
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t->setResponse(m);
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m->deref();
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// notify peer about the changes
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Message* msg = message("call.update");
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msg->addParam("operation","notify");
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msg->addParam("mandatory","false");
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@ -6977,21 +7028,7 @@ void YateSIPConnection::reInvite(SIPTransaction* t)
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putMedia(*msg);
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Engine::enqueue(msg);
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m_reInviting = invite;
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return;
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}
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m_reInviting = invite;
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if (s_refresh_nosdp && !sdp) {
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// be permissive, accept session refresh with no SDP
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updateTarget(t->initialMessage());
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SIPMessage* m = new SIPMessage(t->initialMessage(),200);
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// if required provide our own media offer
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if (!m_rtpForward)
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m->setBody(createSDP());
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t->setResponse(m);
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m->deref();
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return;
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}
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t->setResponse(488);
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return true;
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}
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bool YateSIPConnection::checkUser(SIPTransaction* t, bool refuse)
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@ -8799,6 +8836,7 @@ void SIPDriver::initialize()
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s_multi_ringing = s_cfg.getBoolValue("general","multi_ringing",false);
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s_refresh_nosdp = s_cfg.getBoolValue("general","refresh_nosdp",true);
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s_update_target = s_cfg.getBoolValue("general","update_target",false);
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s_update_verify = s_cfg.getBoolValue("general","update_verify",false);
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s_preventive_bye = s_cfg.getBoolValue("general","preventive_bye",true);
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s_ignoreVia = s_cfg.getBoolValue("general","ignorevia",true);
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s_ipv6 = s_cfg.getBoolValue("general","ipv6_support",false);
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