SER can process any SIP request, doesn't matter the type of method. It
can process the method FOOBAR as long as the whole request is
well-formatted according to SIP specifications (RFC3261). But this is
from a SIP proxy point of view -- the methods request a special behavior
which is specific for SIP clients and SER doesn't do that.
Daniel
On 11/29/04 08:04, Srivathsan S wrote:
Hi folks,
I am trying to find out the methods supported by SER. I am aware that
SER is RFC-3261 compliant, but, do not know to what extent. If someone
can tell me what (in 3261) is supported and what is not, that would be
nice. Also, if SER supports any extensions (3262, 63, 64), please let
me know.
After some Googling, I found that the *basic methods of SIP* are:
*INVITE *– to initiate a session
*Re-INVITE *– if, during a call, either party wants to change the
media; for example to open a video channel
*ACK *– to confirm session establishment and can only be used with INVITE
*BYE *– terminates sessions
*CANCEL *– to cancel a pending INVITE
*OPTIONS *– for capability inquiry
*REGISTER *– to bind a permanent address to current location
*The other methods (found in extensions) of SIP are:*
Other SIP method extensions are defined in *different* RFCs such as:
*SUBSCRIBE *– to subscribe to a service state change. sed for presence
(subscribe to an event and receive
notification), call-back (when other party becomes vailable), voice
mail notification, any event that can be
associated with a trigger (e.g., stock quotes, etc.)
*NOTIFY *– notify a change of service state (e.g., new voice
message).Works in parallel with SUBSCRIBE
*MESSAGE *– for Instant Messaging (user to user messaging). MESSAGE
requests carry the content in the
form of MIME body parts
*REFER *– call transfer
*PUBLISH *– publication of presence information to a server
Does SER support all the above Basic methods? Can someone tell me if
SER supports any of the extensions? Basically, I am trying to use SER
for VoIP integration and I am trying to get all the data I can about
SER for interoperating with various SIP phones. I will not be using
Asterisk for now (as a part of VoIP integration).
Thanks much,
Sri.
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