Indeed, gzcompress is a Kamailio-only concept.
On 17 December 2014 20:58:22 GMT-05:00, Marc Soda <msoda(a)coredial.com> wrote:
So gzcompress is no good with Asterisk then? Is that
meant to be used
only
with another Kamailio proxy?
We're trying to do a WebRTC POC with Kamailio as the proxy. The SIP
headers and SDP are huge! I've never seen such big messages.
Thanks,
Marc
On Wed, Dec 17, 2014 at 6:47 PM, Daniel-Constantin Mierla
<miconda(a)gmail.com
wrote:
On 17/12/14 23:20, Alex Balashov wrote:
> On 12/17/2014 05:14 PM, Marc Soda wrote:
>
>> I'm having a problem reassembling UDP packets on my Asterisk
servers
>> after passing through Kamailio (it
appears to me an OS level
issue,
>> nothing to do with Kamailio). Is there
a way, with Kamailio, to
limit
>> the size of a SIP message? I know I can
just start removing
headers,
>> but that doesn't seem like a
realistic solution. I see that
Kamailio
>> can compress the message body, but can
Asterisk handle that? How
do
other people handle this?
1. Any SIP-compliant endpoint should be able to handle compact
headers. Per RFC 3261 7.3.3 ("Compact Form"):
Implementations MUST accept both the long and short forms of
each header name.
I don't think compact names for headers or joining bodies under
single
header name helps that much, it would be in the
range of few tens of
bytes.
>
> 2. Some headers are critical should not be removed. Others really
are
> mostly useless bloat commonly added by
verbose UACs, and,
practically
> speaking, the other peer will be neither
colder nor warmer if they
are
removed,
unless there is a specific use for them.
Good candidates are:
a) The "Date" header.
b) Accept: headers listing every MIME type in the known universe.
Mentioned on my previous email too -- keep_hf() from textopsx module
can
be handy here.
>
> 3. If one or more of your endpoints offer every codec in the known
> universe in the SDP, you can restrict the codecs offered to reduce
the
SDP size.
Another option to reduce the size -- sdpops module has related
functions
for sdp management.
>
> 4. You could use TCP. In fact, RFC 3261 actually mandates this. Per
> RFC 3261 Section 18.1.1 ("Sending Requests"):
>
> If a request is within 200 bytes of the path MTU, or if it is
larger
> than 1300 bytes and the path MTU is
unknown, the request MUST be
sent
> using an RFC 2914 [43] congestion
controlled transport protocol,
such
> as TCP.
>
> Of course, in reality, nobody cares or follows this, and many SIP
> endpoints don't even support TCP (also mandated by RFC 3261).
>
> 5. In some situations, header bloat comes from requests passing
> through numerous proxies, each of which add a stackable Via header
> and, if applicable, a Route/Record-Route set.
>
> Reducing the number of intermediate proxies can help with this.
>
> 6. You could run the traffic through a lightweight, signalling-only
> B2BUA, such as SEMS, which deals with fragmented UDP in incoming
> requests just fine, but does not reoriginate on leg B all the
bloated
headers
that came in on leg A.
SEMS (like any other application layer program) had very few to do
with
fragmentation. It is the kernel/operating system
that sorts all this.
It
the application is the same
'recvfrom(...)'.
At the end, Asterisk is also a B2BUA and I guess if there is a server
with an OS that can handle udp fragmentation, the Asterisk will be
run
there instead of adding another b2bua.
Cheers,
Daniel
7. Other than these things, there are no real solutions.
-- Alex
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
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Alex Balashov - Principal
Evariste Systems LLC
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Suite 106
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United States
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