[Sipping] Re: I-D ACTION:draft-akhtar-sipping-header-reduction-00.txt

Hello Jonathan,

Thanks for your response. You are right about the details of the
justification of this proposal has been left out from the draft. We can
certainly include them in the next revision. Anyway, here is why Sigcomp
can not solve this problem.

 - To reduce the call setup time in a significant manner the initial
Invite (both originating and terminating) has to have the 
   high compression rate. Sigcomp, however, is not designed to provide
maximum compression in the first set of 
   messages (only message that goes between the UA and the Proxy server
before Invite is the Register message).
   Additionally, the SIP/SDP static dictionary does not have some of the
wireless specific data elements that
   are included in the initial Invite message.

 - Owing to the above facts, an initial SIP Invite can not be reduced to
~210 bytes using Sigcomp approach. We believe
   the proposed draft along with new data elements in the static
dictionary (that are inclusive of the SIP/SDP static
   dictionary as specified by RFC 3485) will accomplish this objective. 
   Please see <draft-akhtar-sipping-3g-static-dictionary-00.txt> 

(http://www.ietf.org/internet-drafts/draft-akhtar-sipping-3g-static-dict
ionary-00.txt)
   for the new data elements for the proposed static dictionary.

 - Using the signaling channel of the wireless access network allows the
SIP session to be initiated simultaneously while
   the RF resources are being acquired. A feature called DoS (Data over
Signaling) in 1xEVDO (the CDMA technology
   of 3GPP2) is a case in point here. Other wireless technologies would
also allow smaller payloads (< 200 bytes) to be 
   transferred over the signaling channel.

 - Assuming very optimistic compression ratio of 50% (by using Sigcomp),
an 1000-byte SIP Invite will be reduced to
   500 bytes. An IMS based initial SIP Invite is typically in the range
of 1200 bytes.

Hope this answers your concern.

Regards,

Haseeb

============================================================

Its not clear from reading this document why everything in here cannot
be accomplished with sigcomp alone. The basic idea, if I understand
right, is that, upon registration, the proxy server stores certain
information about the UA which is not expected to change from request to
request, and it repopulates it when a UA emits a request (which won't
contain that information). That is more or less equivalent to placing
that information into a sigcomp dictionary and then using sigcomp to ask
the server to extract the information from the dictionary.

Thanks,
Jonathan R.
Internet-Drafts at ietf.org wrote:
	A New Internet-Draft is available from the on-line
Internet-Drafts directories.

	Title : New SIP Headers for Reducing SIP Message Size Author(s)
: H. Akhtar, et al.
	Filename : draft-akhtar-sipping-header-reduction-00.txt
	Pages : 18
	Date : 2006-2-28

	Current SIP messages are text based and inherently large,
especially
	when these messages are to be transmitted over the
bandwidth-strained
	wireless access technologies (a typical orginiating SIP Invite
is about 1200 bytes). 

	For most wireless technologies, transmitting the session
initiation messages (such as SIP Invite) over the signaling channel can
reduce the call setup time substantially. The size limitation of these
wireless signaling channels are typically very small (~210 bytes in the
uplink and ~110 bytes in the downlink).

	To address this problem, a new function called Encoding Assitant
(EA) has been introduced in the User Agent (UA) and in the SIP Proxy
server within the network. Additionally, the method provided in this
document defines new SIP option tags and headers. These new option tags
and headers allow the UA and a SIP Proxy server within the network (such
as the P-CSCF), to exchange information using the signaling channels
supported by most wireless access networks.

	A URL for this Internet-Draft is:

http://www.ietf.org/internet-drafts/draft-akhtar-sipping-header-reductio
n-00.txt

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