Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T are two protocols with similar features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of the detail available in the ISUP header. SIP-I was defined by the ITU-T, whereas SIP-T was defined by the IETF. Encryption Concerns about the security of calls via the public Internet have been addressed by encryption of the SIP protocol for secure transmission. The URI scheme SIPS is used to mandate that SIP communication be secured with Transport Layer Security (TLS). SIPS URIs take the form sips:user@example.com. End-to-end encryption of SIP is only possible if there is a direct connection between communication endpoints. While a direct connection can be made via Peer-to-peer SIP or via a VPN between the endpoints, most SIP communication involves multiple hops, with the first hop being from a user agent to the user agent's ITSP. For the multiple-hop case, SIPS will only secure the first hop; the remaining hops will normally not be secured with TLS and the SIP communication will be insecure. In contrast, the HTTPS protocol provides end-to-end security as it is done with a direct connection and does not involve the notion of hops. The media streams (audio and video), which are separate connections from the SIPS signaling stream, may be encrypted using SRTP. The key exchange for SRTP is performed with SDES (), or with ZRTP (). When SDES is used, the keys will be transmitted via insecure SIP unless SIPS is used. One may also add a MIKEY () exchange to SIP to determine session keys for use with SRTP. See also Computer telephony integration (CTI) Computer-supported telecommunications applications (CSTA) H.323 protocols H.225.0 and H.245 IP Multimedia Subsystem (IMS) Media Gateway Control Protocol (MGCP) Mobile VoIP MSCML (Media Server Control Markup Language) Network convergence Rendezvous protocol RTP payload formats SIGTRAN (Signaling Transport) SIP extensions for the IP Multimedia Subsystem SIP provider Skinny Client Control Protocol (SCCP) T.38 XIMSS (XML Interface to Messaging, Scheduling, and Signaling) Notes References External links IANA: SIP Parameters IANA: SIP Event Types Namespace VoIP protocols Videotelephony Application layer protocols