Columbia SIP User Agent CU IRT

User Manual

Features of sipc version 2.72

send INVITE over UDP Yes
receive INVITE over UDP Yes
generate ACK properly Yes
can accept or reject calls Yes
SDP with single m and c line, one codec Yes
To, From, Call-ID, CSeq, Via, Content-Length, Content-Type headers handled properlyYes
generate tags in To field Yes
send basic call termination with BYE via UDP Yes
receive BYE over UDP Yes
compact form for headers Yes
reject unknown request methods with 501 response Yes
send/receive RTP media, possibly without RTCP Use external media application
support TCP for all messages Yes
Require, Proxy-Require No
handle packet loss for INVITE and BYE (with exponential backoff)Yes
pays attention to Contact header in INVITE and in 2xx response to INVITE (i.e., goes directly to peer for following requests)Yes
process CANCEL for INVITE Yes
Authentication for registrations: basic Yes
Authentication for registrations: digest Yes
allow redirection to web pages or email Yes
receive text or HTML in 3xx or 4xx responses No
Accept headers without SDP No
DNS SRV recordsYes
non-gateways: register with periodic refresh to unicast address, paying attention to Expires header in REGISTER responseYes
understands redirection Yes
multiple codecs listed in SDP m line, finds common one with peerYes
multiple SDP m= lines handled correctly Yes
unknown SDP m= media types handled correctly (i.e., rejected with port 0)Yes
Domain name as well as IP address accepted in SDP c= lineYes
generate RTCP packets Use external media application
respond to OPTIONS request Yes
allows non-SIP URLs in REGISTER Yes
copy Record-Route from response into Route of request and route appropriatelyYes
checks equality of action parameters on REGISTERNo
can retrieve current registrations Yes
can clear registrations with Contact: * and Expires: 0Yes
automatically tries redirections (recursing UA) Yes
generate multicast REGISTER No
re-INVITE: suspending a stream Yes
re-INVITE: resuming a stream Yes
re-INVITE: closing single stream Yes
re-INVITE: changing codecs No
re-INVITE: add a stream Yes
re-INVITE: change media address to different address or port (mobility)Yes
send text or HTML in 3xx and 4xx responses No
Expires for INVITE No
third party registration Yes
generate tel: URL request and proxy them to designated serverNo
process MIME multipart responses Yes
beyond RFC3261
Emergency call handling Yes
INFO method Yes
TRANSFER method Yes
183 response No
caller preferences No
tones (DTMF) in RTP No
SIP for presence (SUBSCRIBE and NOTIFY method) Yes
SIP for instant message (MESSAGE method) Yes
SIP for device control (DO metod) Yes
Use MESSAGE method for web browsing sharing Yes
Send location information in MIME Yes
service engine
Language for End System Services (LESS) scripts Yes
Service creation environment Supports both graphical and web based
Service learning (Automatically generate service scripts) Yes
Feature interaction handling Yes
Address book Yes
Session Announcement Protocol support Yes
Real Time Streaming Protocol support Yes
Service Location Protocol support Yes
Location sensing Yes

Last updated by Xiaotao Wu