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.

Yes, I suppose I CAN switch to TCP, but I don't WANT to switch to TCP.

My main gripe with TCP is that it holds a socket open for every registered endpoint.  If I 
have to restart a profile, I have to leave it down for several minutes until all the 
sockets close before I can start it up again.

I don't suppose there is a way to disable the <local> and <remote> tags in the body?

Thanks!



Ken Rice wrote:

Switch to TCP... This is why the RFCs tell you to support both UDP and TCP
for SIP... 


On 1/9/13 9:36 AM, "Tim St. Pierre" <fs-list@communicatefreely.net> wrote:

Hello,

I'm having a problem with BLF in some cases, and I found that the length of
the username 
was a factor in whether the lamps worked or not.

When I did some packet captures, I found that problem packets were 1500 bytes
or so. 
Since the endpoint is on a DSL line that has an MTU of 1460, the packet
doesn't fit.

I'm already using compressed headers.  Is there a way to reduce the packet
size more, or 
is there another solution?  I will include an example below:  Note that my IP
address and 
domain substutions may change the packet size a little, but the point is the
same.

Thanks!

NOTIFY sip:XXXX@YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0
v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp
Route:<sip:aaa.aaa.aaa.aaa:61906>;transport=udp
Max-Forwards:70
f:<sip:accountcode-701@ourdomain.com>;tag=TrmHE4FBIrB4
t:"Tim St. Pierre"<sip:XXXX@ourdomain.com>;tag=ecc2498bfd
i:3decd12decccafe1
CSeq:448376002 NOTIFY
m:<sip:accountcode-701@xxx.xxx.xxx.xxx:5070;transport=udp;fs_nat=yes;fs_path=s
ip:yyy.yyy.yyy.yyy:61906%3Btransport%3Dudp>
User-Agent:Communicate Freely 2.0
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,
PUBLISH,SUBSCRIBE
k:timer,precondition,path,replaces
o:dialog
u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-sessio
n-description,presence.winfo,message-summary,refer
Subscription-State:active;expires=1137
c:application/dialog-info+xml
l:639

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="477"
state="partial" 
entity="sip:accountcode-701@ourdomain.com">
<dialog id="accountcode-701" direction="initiator">
<state>confirmed</state>
<local>
<identity 
display="park">sip:accountcode-701@ourdomain.com;proto=park</identity>
<target uri="sip:accountcode-701@ourdomain.com;proto=park">
<param pname="+sip.rendering" pvalue="no"/>
</target>
</local>
<remote>
<identity display="park">sip:accountcode-701</identity>
<target

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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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