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Today's Topics:
1. Re: Ringing / Session Progress (180/183) (Anthony
Orlando)
2. O line of SDP (mwilliam prusty)
3. Re: O line of SDP (Herve Jourdain)
4. Re: Ringing / Session Progress (180/183) (Ramesh
Chauhan)
------------------------------------------------------------
----------
Message: 1
Date: Thu, 10 Jan 2008 01:07:31 -0800 (PST)
From: Anthony Orlando <avorlandoyahoo.com>
Subject: Re: [SIPForum-discussion] Ringing / Session
Progress
(180/183)
To: Donald Lee <baolovebaogmail.com>, Ramesh
Chauhan
<chauhan_delhiyahoo.com>
Cc: discussionsipforum.org
Message-ID: <898609.6601.qmweb51007.mail.re2.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Partially correct. The key is whether or not there is
SDP. The old rule of thumb was that if SDP is present
then listen inband for ringing, tones, or
announcements. If no SDP then phone provides ring
tone. This rule has somewhat changed by RFC3960.
--- Donald Lee <baolovebaogmail.com> wrote:
> I think you are partly right.
> In many scenarios that after receiving the 183
> response and doing a
> PRACK procedure, another 180 response arrive. In
> this case, if the RTP
> stream is also sending back, then the originator
> needn't generate
> local ring back tone, otherwise it should play
> itself.
>
> But according to my understanding and the
> practicality, no matter 183
> or 180 responses that received, the UAC should not
> generate the ring
> back tone if there is RTP stream incoming. It's
> color ring back tone.
>
> On Jan 10, 2008 7:04 AM, Ramesh Chauhan
> <chauhan_delhiyahoo.com> wrote:
> > Hi,
> >
> > Can you please tell me the diference between
> Ringing(180) and Session
> > progress(183).
> >
> > As per me, if we recieve Session Progress(183),
> then our media will start
> > flowing from farend to us and from us to farend.
> and if we receive
> > Ringing(180) then our device will generate his
own
> ring on the phone.
> > Eg:
> > Session Progress(183):
> > Callee SIPProxy
> Caller
> > ----------Invite--------------->
> >
> ----------Invite--------------->
> >
> <--------Trying--------------
> > <--------Trying--------------
> >
> <--SessioProgress(183)-
> > <--SessioProgress(183)-
> > --------Here we will receive RTP
> from-------------------
> >
> > Ringing (180):
> > Session Progress(183):
> > Callee SIPProxy
> Caller
> > ----------Invite--------------->
> >
> ----------Invite--------------->
> >
> <--------Trying--------------
> > <--------Trying--------------
> >
> <--ringing(183)------------
> > <--ringing(183)------------
> > --------Here our device will generate local
> ring-----------
> >
> > Am I correct ???????????/
> >
> >
> > Regards
> > Chauhan
> >
> >
> > ________________________________
> > Looking for last minute shopping deals? Find them
> fast with Yahoo! Search.
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> > http:
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> > Post to the list at discussionsipforum.org
> >
> >
>
>
>
> --
> BR
> Donald
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
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------------------------------
Message: 2
Date: 10 Jan 2008 09:43:36 -0000
From: "mwilliam prusty" <callwilliam90653rediffmail.com>
Subject: [SIPForum-discussion] O line of SDP
To: discussionsipforum.org
Cc: william.prustygmail.com
Message-ID: <20080110094336.607.qmailwebmail69.rediffmail.com>
Content-Type: text/plain; charset="iso-8859-1"
?
Hi
can any body tell me suppose in the SDP of the INVITE
contain o=root 379 379 IN IP4 <IP Address> . The SDP
of 200 ok will conatin
o= root 379 379 IN IP4 <IP address>. Here my attention
is to check whetehr in the O line of SDP of 200 ok will
conatin 379 379 , or 380 380 or it can be anything ??
william
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Message: 3
Date: Thu, 10 Jan 2008 11:55:44 +0100
From: "Herve Jourdain" <herve.jourdainmstarsemi.com>
Subject: Re: [SIPForum-discussion] O line of SDP
To: "'mwilliam prusty'" <callwilliam90653rediffmail.com>,
<discussionsipforum.org>
Cc: william.prustygmail.com
Message-ID: <200801101055.m0AAtfDM097353mailsqr.mstarsemi.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I think that if it's the answer to the INITIAL INVITE, the
"session id" and
"version" fields of the 200 OK will NOT be
correlated to the ones of the
INVITE.
So I think that if in INVITE you have:
o=root 379 379 IN IP4 <IP1>,
You will have in 200 OK:
o=foo xxx yyy IN IP4 <IP2>
As far as I know, xxx can be different from yyy, even if the
implementations
I've seen usually use the same for both (as it's recommended
it should be a
NTP timestamp.).
Also, xxx and yyy would/could be different from 379 in your
example.
Now, for modifications to the SDP in the middle of the
dialog, each party
should send the offer - and response - based on the initial
"session id" and
"version" fields they initially sent, with one
additional rule:
If there has been ANY modification to the SDP - and if you
send again, it's
probably the case - the "session id" remains the
same, but the "version"
field must be incremented by 1.
Regards,
Herve
_____
From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
mwilliam prusty
Sent: jeudi 10 janvier 2008 10:44
To: discussionsipforum.org
Cc: william.prustygmail.com
Subject: [SIPForum-discussion] O line of SDP
Hi
can any body tell me suppose in the SDP of the INVITE
contain o=root 379
379 IN IP4 <IP Address> . The SDP of 200 ok will
conatin
o= root 379 379 IN IP4 <IP address>. Here my attention
is to check whetehr
in the O line of SDP of 200 ok will conatin 379 379 , or
380 380 or it
can be anything ??
william
<http://adworks.rediff.com/cgi-bin/AdW
orks/click.cgi/www.rediff.com/signatur
e-home.htm/1050715198Middle5/2027367_2020178/2027187/1?PARTNER=3&OAS_Q
UERY=
null> ICICI PRU
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Message: 4
Date: Thu, 10 Jan 2008 03:17:19 -0800 (PST)
From: Ramesh Chauhan <chauhan_delhiyahoo.com>
Subject: Re: [SIPForum-discussion] Ringing / Session
Progress
(180/183)
To: dwardithinktest.com, discussionsipforum.org
Message-ID: <1742.59977.qmweb34406.mail.mud.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I ave got many replies on this, thanks to all.
SIPMedia Proxy Server is receiving media packets from the
callee ater getting 183, but it is not sending these Media
packets to the Caller. Due to which the caller is not able
to hear the Ring on his Phone..
Some one told me that "MediaProxy will forward the
RTP received from the carrier only if the devices sends
first packet to the MediaProxy during 183, If the device is
not going to send the packet during 183, we wont be able to
identify the Source IP and Port of the caller
device/Softphone to which we have to send the media."
And in my case, I have captured the ethreal logs and found
that the caller device is sending first packet after getting
200-OK.
So, what should we do (in configuration of device), so
that it will send first packet after getting 183....
Any reply will be very helpful for me.....
Regards
Chauhan
dwardithinktest.com wrote:
v
{behavior:url(#default#VML);} o
{behavior:url(#default#VML);} w
{behavior:url(#default#VML);} .shape
{behavior:url(#default#VML);} Hi Ramesh ?
Your personal explanation or physical difference is correct
for 180 vs 183. The ?180 Ringing? response is used
primarily to indicate that the INVITE has been received by
he user agent and that alerting is taken place. This is
especially important when interfacing with PSTN ntwks where
the ISUP address Complete Msg (ACM) is mapped. However no
media is passed. ?183 ? Session Progress? indicates that
information about the progress of the session or call state
may be presented in a msg body or media stream. A typical
use of this response is to allow the UAC to hear media (i.e.
ring tone, busy tone, or a recorded msg via the bearer
channel in calls through a gateway into the PSTN. This is
bec call progress info is carried in the media stream in the
PSTN as you may know. A great book that you
should have in your library is ?SIP understanding the
Session Information Protocol? 2nd Edition by Alan B.
Johnston ? ISBN 1-58053-655-7. Btw, an excellent tool to
generate/capture sip responses for pre/post deployment
purposes is the iThinkTest Toolsets. Let me know if you
would like to try them out? ~ dw
---------------------------------
From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
Ramesh Chauhan
Sent: Wednesday, January 09, 2008 3:04 PM
To: discussionsipforum.org
Subject: [SIPForum-discussion] Ringing / Session Progress
(180/183)
Hi,
Can you please tell me the diference between
Ringing(180) and Session progress(183).
As per me, if we recieve Session Progress(183), then our
media will start flowing from farend to us and from us to
farend. and if we receive Ringing(180) then our device will
generate his own ring on the phone.
Eg:
Session Progress(183):
Callee SIPProxy
Caller
----------Invite--------------->
----------Invite--------------->
<--------Trying--------------
<--------Trying--------------
<--SessioProgress(183)-
<--SessioProgress(183)-
--------Here we will receive RTP
from-------------------
Ringing (180):
Session Progress(183):
Callee SIPProxy
Caller
----------Invite--------------->
----------Invite--------------->
<--------Trying--------------
<--------Trying--------------
<--ringing(183)------------
<--ringing(183)------------
--------Here our device will generate local
ring-----------
Am I correct ???????????/
Regards
Chauhan
---------------------------------
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with Yahoo! Search.
with regards
Ramesh Chauhan
---------------------------------
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