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Statements (89)
| Predicate | Object |
|---|---|
| gptkbp:instanceOf |
gptkb:network_protocol
|
| gptkbp:access |
Digest authentication
|
| gptkbp:category |
gptkb:network_protocol
gptkb:signaling_protocol |
| gptkbp:definedIn |
gptkb:RFC_3261
|
| gptkbp:encryption |
gptkb:SRTP
TLS |
| gptkbp:extendsTo |
SIP extensions
|
| gptkbp:firstPublished |
2002
|
| gptkbp:format |
text-based
|
| gptkbp:fullName |
gptkb:Session_Initiation_Protocol
|
| gptkbp:headerField |
gptkb:event
gptkb:road gptkb:Via Contact Authorization Content-Length Content-Type Expires User-Agent Allow Allow-Events CSeq Call-ID From Max-Forwards Min-SE P-Access-Network-Info P-Asserted-Identity P-Asserted-Service P-Associated-URI P-Called-Party-ID P-Charging-Function-Addresses P-Charging-Vector P-Early-Media P-Media-Authorization P-Preferred-Identity P-Preferred-Service P-Profile-Key P-Served-User P-User-Database P-Visited-Network-ID Proxy-Authenticate Proxy-Authorization Record-Route Referred-By Replaces Require Session-Expires Subscription-State Supported To WWW-Authenticate |
| gptkbp:homeport |
5060
5061 |
| gptkbp:relatedTo |
gptkb:H.323
gptkb:MGCP gptkb:SIP_Trunking |
| gptkbp:standardizedBy |
gptkb:IETF
true |
| gptkbp:supports |
gptkb:NAT_traversal
instant messaging video calls voice calls call forwarding call hold call management call setup call transfer conference calls presence information user availability user location |
| gptkbp:supportsProtocol |
gptkb:TCP
gptkb:UDP TLS |
| gptkbp:usedBy |
gptkb:IP_PBX
softphones SIP phones VoIP gateways |
| gptkbp:usedFor |
controlling multimedia communication sessions
signaling |
| gptkbp:usedIn |
gptkb:VoIP
instant messaging video conferencing |
| gptkbp:bfsParent |
gptkb:IETF
gptkb:WebRTC |
| gptkbp:bfsLayer |
4
|
| https://www.w3.org/2000/01/rdf-schema#label |
SIP
|