[Freeswitch-users] SIP packet exceeds MTU?

Tim St. Pierre fs-list at communicatefreely.net
Wed Jan 9 19:53:56 MSK 2013


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 at 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 at 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 at ourdomain.com>;tag=TrmHE4FBIrB4
>> t:"Tim St. Pierre"<sip:XXXX at ourdomain.com>;tag=ecc2498bfd
>> i:3decd12decccafe1
>> CSeq:448376002 NOTIFY
>> m:<sip:accountcode-701 at 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 at ourdomain.com">
>> <dialog id="accountcode-701" direction="initiator">
>> <state>confirmed</state>
>> <local>
>> <identity 
>> display="park">sip:accountcode-701 at ourdomain.com;proto=park</identity>
>> <target uri="sip:accountcode-701 at 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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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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> 




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