[SIPForum-discussion] confussion about use of tcp/udp
Manuel Stein
manuel.stein at web.DE
Mon Jun 9 21:52:51 UTC 2008
Hi all,
I've listened to a few people who prefer UDP over TCP. As a matter of
fact, TCP has all these downsides like slow start, connect, soo
difficult to implement, etc. *g* but hey, TCP stacks are available and
if you find yourself a nice configurable one, you can easily pimp the
network up.
But I'm really interested to know whether talking about TCP overhead and
using an every-script-kiddie-aged-8-can-read-this
why-not-adding-a-few-more-superfluous-headers
oh-let's-make-it-ambiguous-by-allowing-every-information-to-be-attached-everywhere
ASCII-protocol at the same time isn't somehow contradicting in itself ^.^
However, regarding the question, the answer is: yes
The initial request might be below MTU size and is sent over UDP. When
Vias, Record-Routes or P-HEaders, etc. add up, at some point it might
come to TCP. The responses are processed backwards consuming Vias and
the transport is the same for all responses. Note that for subsequent
requests of a dialog, the rule applies independently.
You may try to fragment packets, but this would be more likely a "SIP
profile" *g*, because as Francois correctly assumed and
Rosenberg&Schulzrinne already noticed is that it doesn't work out there,
because people ever since only implemented half the standards and
thought that the last quantile wouldn't add any benefit.
Best regards,
(sorry for the sarcasm - well, I like it)
Manuel
Sandeep Mohapatra wrote:
> Hello guys,
>
> In SIP the request can go over TCP or UDP. However most of the stacks
> use UDP as default transport protocol because it has less bytes(8
> bytes) and hence less traffic issues in compared to TCP(12 to 20 bytes).
>
> However you can specifically implement the transport protocol by
> adding one parameter on the URI.
> for example:
> <sip:1233 at 10.250.200.50:5060;transport=UDP>
> <sip:1233 at 10.250.200.50:5060;transport=udp>
> <sip:1233 at 10.250.200.50:5060;transport=TCP>
> <sip:1233 at 10.250.200.50:5060;transport=tcp>
>
> Hope this clarify the issues.
>
> Regards,
> Sandeep
>
> --- On *Mon, 9/6/08, Francois Audet /<audet at nortel.com>/* wrote:
>
> From: Francois Audet <audet at nortel.com>
> Subject: Re: [SIPForum-discussion] confussion about use of tcp/udp
> To: "Samir Patni" <samir_patni at persistent.co.in>, "Neill
> Wilkinsonj" <neill.wilkinson at quortex.com>, khalikh at yahoo.com,
> discussion at sipforum.com
> Cc: "kranti kumar" <kkril at cooltoad.com>
> Date: Monday, 9 June, 2008, 9:53 PM
>
> It pretty much doesn't work like this in real life.
>
> You pick either TCP or UDP and in the case of UDP, pray that you
> won't get fragmentation problems.
>
> ------------------------------------------------------------------------
> *From:* discussion-bounces at sipforum..org
> [mailto:discussion-bounces at sipforum.org] *On Behalf Of *Samir
> Patni
> *Sent:* Monday, June 09, 2008 04:48
> *To:* 'Neill Wilkinsonj'; khalikh at yahoo.com;
> discussion at sipforum.com
> *Cc:* 'kranti kumar'
> *Subject:* Re: [SIPForum-discussion] confussion about use of
> tcp/udp
>
>
>
> Hi All,
>
>
>
> Can a call contain both TCP and UDP messages.
>
>
>
> From section 18.1.1 from rfc 3261
>
> “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.”
>
>
>
> Consider a scenario in which Invite is sent over UDP and for
> some other subsequent request/response (say A).
>
> Within the same call the Request length is more then 1300
> bytes. Should this request A be sent over
>
> TCP?
>
>
>
> Has someone seen two transport layer protocols used in the
> same call?
>
>
>
> Thanks,
>
> Samir Patni
>
> ------------------------------------------------------------------------
>
> *From:* discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] *On Behalf Of *Neill
> Wilkinsonj
> *Sent:* Monday, June 09, 2008 12:45 PM
> *To:* khalikh at yahoo.com; discussion at sipforum.com
> *Cc:* 'kranti kumar'
> *Subject:* Re: [SIPForum-discussion] confussion about use of
> tcp/udp
>
>
>
> *RTP (voice) is always carried over UDP. There is one notable
> exception and I believe Skype can sometimes use TCP to carry
> voice. TCP generally has some undesirable properties
> (Head-of-Line blocking and long timers related to connection
> state), which make it unsuitable for carry realtime traffic
> such as voice.*
>
> * *
>
> *SIP signalling can travel over UDP or TCP, generally the
> UA(C) will choose one or the other at the start of the
> interaction and the UA(S) will respond using the same for the
> complete dialogue. That is my observation in Practice.*
>
> * *
>
> *Someone else on the list more conversant with RFC which to
> confirm or clarify?*
>
> * *
>
> * Neill....;o)*
>
> *From:* discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] *On Behalf Of
> *khalikh uddin
> *Sent:* 09 June 2008 07:18
> *To:* discussion at sipforum.com
> *Cc:* kranti kumar
> *Subject:* [SIPForum-discussion] confussion about use of tcp/udp
>
>
>
> hai all
> can anybody tell me
> *when a sip enabled end device opens a TCP connection and
> when it uses UDP to send *REQ/RESPONSES to proxy or other UA.
> *can it use both in a aparticular diloge i.e to send INVITE
> use TCP and to send ACK use
> UDP.
> *Can we force it to use TCP/UDP.
> *For media transmission SIP uses only tcp.Is this true.
>
>
>
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