SIP packet exceeds MTU?
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=sip: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-session-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
Unfortunately if you're using compressed headers then the packet's pretty much as small as it can get. Can you use TCP as the transport for SIP? If so that should handle the fragmentation ok. -Steve On 9 January 2013 15:36, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
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.
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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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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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You don't have to agree to switch. The specs says you should 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. On 01/10/2013 12:53 AM, Tim St. Pierre wrote:
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Joegen Baclor
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Ken Rice
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Steven Ayre
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Tim St. Pierre