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Added code and setting to prevent rebuilding the RTP when only the remote port has changed in the SDP offer. 

This can prevent a neverending sequence of reINVITEs, each end trying to adjust to the changes of the other.


git-svn-id: http://voip.null.ro/svn/yate@2786 acf43c95-373e-0410-b603-e72c3f656dc1
This commit is contained in:
paulc 2009-08-12 15:18:36 +00:00
parent 4e9abe5c08
commit 71f34fd5cf
2 changed files with 9 additions and 3 deletions
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4
conf.d/ysipchan.conf.sample
View File

@ -200,3 +200,7 @@
; 1xx_change_formats: bool: Provisional messages can change the formats list
;1xx_change_formats=yes
; ignore_sdp_port: bool: Ignore SDP changes if only the port is different
; This allows preserving the local RTP session and port
;ignore_sdp_port=no

8
modules/ysipchan.cpp
View File

@ -168,10 +168,10 @@ public:
{ return m_lCrypto; }
inline bool securable() const
{ return m_securable; }
inline bool sameAs(const NetMedia* other) const
inline bool sameAs(const NetMedia* other, bool ignorePort = false) const
{ return other && (other->formats() == m_formats) &&
(other->transport() == m_transport) &&
(other->remotePort() == m_rPort); }
(ignorePort || (other->remotePort() == m_rPort)); }
const char* fmtList() const;
bool update(const char* formats, int rport = -1, int lport = -1);
void update(const Message& msg, bool pickFormat);
@ -655,6 +655,7 @@ static bool s_forward_sdp = false;
static bool s_start_rtp = false;
static bool s_ack_required = true;
static bool s_1xx_formats = true;
static bool s_ignore_port = false;
static bool s_auth_register = true;
static bool s_reg_async = true;
static bool s_multi_ringing = false;
@ -2585,7 +2586,7 @@ void YateSIPConnection::setMedia(ObjList* media)
for (; l; l = l->skipNext()) {
NetMedia* m = static_cast<NetMedia*>(l->get());
// preserve data endpoints if media didn't change
if (m->sameAs(static_cast<NetMedia*>((*media)[*m])))
if (m->sameAs(static_cast<NetMedia*>((*media)[*m]),s_ignore_port))
continue;
clearEndpoint(*m);
}
@ -5207,6 +5208,7 @@ void SIPDriver::initialize()
s_reg_async = s_cfg.getBoolValue("registrar","async_process",true);
s_ack_required = !s_cfg.getBoolValue("hacks","ignore_missing_ack",false);
s_1xx_formats = s_cfg.getBoolValue("hacks","1xx_change_formats",true);
s_ignore_port = s_cfg.getBoolValue("hacks","ignore_sdp_port",false);
initAudioCodecs();
if (!m_endpoint) {
m_endpoint = new YateSIPEndPoint();