Re: [Sip] Route header and transport

["Attila Sipos" <attila.sipos at vegastream.com>]

Technically, you can send the ACK with either TCP or UDP.

>From RFC 3261, you can use UDP upto a certain message size:

   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. If this causes a change in the transport protocol from the
   one indicated in the top Via, the value in the top Via MUST be
   changed.  This prevents fragmentation of messages over UDP and
   provides congestion control for larger messages.  However,
   implementations MUST be able to handle messages up to the maximum
   datagram packet size.  For UDP, this size is 65,535 bytes, including
   IP and UDP headers.




-----Original Message-----
From: sip-bounces at ietf.org [mailto:sip-bounces at ietf.org] On Behalf Of
Andrey Kovalenko
Sent: 31 July 2008 08:35
To: sip at ietf.org
Subject: [Sip] Route header and transport

Hello.

I have following problem:

For example, UAC decides to send INVITE via TCP transport and does it
like so:

INVITE sip:echo at sipnet.ru SIP/2.0
Call-ID: e271e3d45ca69c0a
CSeq: 1 INVITE
From: "user" <sip:test at 192.168.1.4>;tag=1e5f7ab5
To: <sip:echo at sipnet.ru>
Via: SIP/2.0/TCP
192.168.1.4:5060;branch=z9hG4bKce32ffc97438620028ba85a78311d3cb
Max-Forwards: 70
Contact: <sip:test at 192.168.1.4:5060>
Content-Type: application/sdp
Content-Length: 227

Then OK is receivied and it looks like so

SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.1.4:5060;branch=z9hG4bKce32ffc97438620028ba85a78311d3cb;received
=192.168.1.4
Record-Route:
<sip:88447-212.53.40.72.dialog.cgatepro;lr>,<sip:212.53.40.72:5060;lr>,<
sip:212.53.40.40:5060;lr>
From: "user" <sip:test at 192.168.1.4>;tag=1e5f7ab5
To: <sip:echo at sipnet.ru>;tag=000000000518036-57737A9D_lnqxjfk-660DE7A8
Call-ID: e271e3d45ca69c0a
CSeq: 1 INVITE
Contact: <sip:signode-lnqxjfk-518036-660DE7A8 at 212.53.40.40>
Supported: 100rel,timer,replaces,histinfo
User-Agent: CommuniGatePro-callLeg/5.2.6
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
Content-Type: application/sdp
Content-Length: 193

Then ACK is sent to URI that is defined in Contact header:

ACK sip:signode-lnqxjfk-518036-660DE7A8 at 212.53.40.40;transport=tcp
SIP/2.0
Via: SIP/2.0/TCP
192.168.1.4:5060;branch=z9hG4bKca9dbdeb9f803d0cdba2c2124abd9440
CSeq: 1 ACK
From: "user" <sip:test at 192.168.1.4>;tag=1e5f7ab5
To: <sip:echo at sipnet.ru>;tag=000000000518036-57737A9D_lnqxjfk-660DE7A8
Call-ID: e271e3d45ca69c0a
Supported: 100rel,timer,replaces,histinfo
User-Agent: CommuniGatePro-callLeg/5.2.6
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
Max-Forwards: 70
Route:
<sip:212.53.40.40:5060;lr>,<sip:212.53.40.72:5060;lr>,<sip:88447-212.53.
40.72.dialog.cgatepro;lr>
Contact: <sip:test at 192.168.1.4:5060>
Content-Length: 0


Should this ACK be sent via TCP or UDP, whether URI in Route header does
not explicitly defines transport?



--
Best regards,
 Andrey                          mailto:akvakh at gmail.com

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sipping at ietf.org for new developments on the application of sip