SIP ProtocolOperation

1.What does the [H14.17] in RFC
2543 stand for?


2.Do callers need to know the
location of the location server?

3.Which parts of SIP are case-
sensitive or case-insensitive?

4.What is the difference between a
call leg and a call id?

5.What is the difference between
tag and branch-id?

6.How can one recognize a
retransmitted request?

7.How does a caller find its local
registrar?

8.Is the domain of the request-URI
and the To header always the
same?

9.Are ACK requests retransmitted?

10.How are BYE requests routed?

11.Can I CANCEL requests other
than the first INVITE?

12.What is the relationship between
the From, Contact, Via and
Record-Route/Route headers?

13.How are URLs compared?

14.What's the difference between
the request URIs tel:
+12125551212 and
sip:12125551212@gw.com?

15.Does SIP do admission control?

16.Does SIP administer bandwidth?

17.Do I always need a proxy or
redirect server?

18.How does a caller find its proxy
server?

19.What's the difference between a
stateless and a stateful proxy
server?

20.Why can a forking SIP proxy not
be stateless?

21.How do I handle the case where
multiple requests are received at
a SIP server, each from a
different upstream proxy?

22.How does a caller find the
remote SIP client of the callee?

23.How does SIP get through a
firewall?

24.What are the issues if SIP is used
behind a NAT?

25.How does SIP do "call progress
tones" or "ring back"?

26.Does SIP do keep-alive?

27.Why does SIP not have a
Content-Transfer-Encoding
header?

28.I want SIP to be more compact.
What can I do?