Add some notes on problems with the TCP/TLS implementation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -32,16 +32,27 @@
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* \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
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* settings, dialplan commands and dialplans apps/functions
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*
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*
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* TODO:s
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* \todo Better support of forking
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* \todo VIA branch tag transaction checking
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* \todo Transaction support
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* \todo We need to test TCP sessions with SIP proxies and in regards
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* to the SIP outbound specs.
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* ******** TCP implementation changes needed
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* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
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* \todo Save TCP/TLS sessions in registry
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* \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
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* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
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* The tcpbindaddr config option should only be used to open ADDITIONAL ports
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* \todo Be prepared for one outbound and another incoming socket per pvt. This applies
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* specially to communication with other peers (proxies).
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* \todo We need to test TCP sessions with SIP proxies and in regards
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* to the SIP outbound specs.
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* \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
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* \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
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* message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
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* \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
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* multiple domains in our TLS implementation, meaning one socket and one cert per domain
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*
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*
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* ******** General TODO:s
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* \todo Better support of forking
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* \todo VIA branch tag transaction checking
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* \todo Transaction support
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*
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* \ingroup channel_drivers
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*
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@ -647,7 +658,7 @@ static const struct cfsip_options {
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/*! \brief SIP Methods we support
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\todo This string should be set dynamically. We only support REFER and SUBSCRIBE is we have
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\todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
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allowsubscribe and allowrefer on in sip.conf.
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*/
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#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
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