[asterisk-users] Realtime & sip.conf

[hugolivude]

Hi -

I'm looking into realtime and I'm having a bit of a problem with the SIP part.

My review of the posts seems to indicate that I should use realtime
static for the [general] part of my sip.conf including the
registration commands:

   register=><did>:<secret>@<domain>/<did context>

and use realtime realtime (funny name!) for peers and friends:

[myprovider]
type=peer
auth=md5
username=...
fromuser=...
fromdomain=...
secret=...
host=...
port=5060
nat=yes
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
context=incoming-sip

Is this correct?  What's throwing me off is this statment found @
http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static:

   NOTE: You can only store a static config OR a RealTime config. You
cannot, for example, store
              sip.conf and use sipfriends via RealTime.

If I am correct, it would suggest that I'll have to do a reload when I
add a DiD, but a reload won't be necessary if a new SIP client is
added.  Do I have it right?

Also, what's the difference between a peer and a user?  I used to
think that a "user" was an agent  authorized to call in to my * box, a
"peer" was an agent I could reach and a "freind" was both.  What's
throwing me off now is the statement found @
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966:

    With newer versions of Asterisk the concept of SIP 'users' will be
phased out.

I can't understand this especially in the context of extconfig.conf
that uses both a sipuser and sippeer entry.  Could someone clarify for
me?

Thanks,
H