Dear
all,
Some 'almost newbie'
questions about SIP (user agent side). Thank you very much in advance for your
response. In addition to the group, direct response to [EMAIL PROTECTED] will
be appreciated.
1- Why the minimum
implementation of a UA does not require the 'bye' support ? How will be the
session terminated, using 'minimum' user agents ?
2- Should a
terminal, which is intended to transmit calls to the PSTN, support
something from TRIP (Telephony Routing Over IP) or is it only the problem of the
gateway ?
3- When deploying
user agents on the network, how are they configured, concerning proxies ? Can we
specify several proxys or just one ? I kow that the user agent will first
contact a server having the same hostname, and then a outbond proxy
configured manually. But is it possible to specify several proxys of differnt
types (redirect, proxys, etc...)
4- Does somebody
know where to find a comprehensive text about the way of placing conferences,
using SIP ?
5- In a SIP
session, at what time should the User Agent be able to listening to
the RTP port ? Once the session is established (invite, ok 200, ack) or at
any time during the message exchanges ?
Again, thank
you,
Catherine
Marselli
