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[saurabh_aggarwal at ho...]

I am having an *odd* issue, which i am not sure freeswitch is to be blamed for.

Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.1?

I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch.

Any suggestions?

-Saurabh


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[saurabh_aggarwal at ho...]

Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?).

But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away?

-Saurabh


From: saurabh_aggarwal@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Tue, 18 Nov 2008 07:34:11 +0000
Subject: [Freeswitch-users] SIP Invite IP fragmentation issue

.ExternalClass .EC_hmmessage P {padding:0px;} .ExternalClass body.EC_hmmessage {font-size:10pt;font-family:Verdana;} I am having an *odd* issue, which i am not sure freeswitch is to be blamed for.

Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.1?

I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch.

Any suggestions?

-Saurabh





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[ibc at aliax.net]

2008/11/18 Saurabh Aggarwal <saurabh_aggarwal@hotmail.com>:

[krice at suspicious.org]

Its not really possible other then enabling compact headers or by getting rid of codecs that you don’t actually want to use... Another thing you could do is get your broken ISP to fix their firewall... It is not correct to just drop fragmented packets just because they are fragmented.. This is something that will happen on a regular basis on the internet as not everything has an MTU of 1500


From: Saurabh Aggarwal <saurabh_aggarwal@hotmail.com>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Tue, 18 Nov 2008 10:19:55 +0000
To: <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue

Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?).

But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away?

-Saurabh


From: saurabh_aggarwal@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Tue, 18 Nov 2008 07:34:11 +0000
Subject: [Freeswitch-users] SIP Invite IP fragmentation issue

I am having an *odd* issue, which i am not sure freeswitch is to be blamed for.

Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.1?

I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch.

Any suggestions?

-Saurabh





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[saurabh_aggarwal at ho...]

enabling compact headers - what is that?

-Saurabh



Date: Tue, 18 Nov 2008 04:29:28 -0600
From: krice@suspicious.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue

Its not really possible other then enabling compact headers or by getting rid of codecs that you don’t actually want to use... Another thing you could do is get your broken ISP to fix their firewall... It is not correct to just drop fragmented packets just because they are fragmented.. This is something that will happen on a regular basis on the internet as not everything has an MTU of 1500




From: Saurabh Aggarwal <saurabh_aggarwal@hotmail.com>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Tue, 18 Nov 2008 10:19:55 +0000
To: <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue

Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?).

But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away?

-Saurabh




From: saurabh_aggarwal@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Tue, 18 Nov 2008 07:34:11 +0000
Subject: [Freeswitch-users] SIP Invite IP fragmentation issue

I am having an *odd* issue, which i am not sure freeswitch is to be blamed for.

Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.1?

I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch.

Any suggestions?

-Saurabh







Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here <http://clk.atdmt.com/MRT/go/119462413/direct/01/>



Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here <http://clk.atdmt.com/MRT/go/119462413/direct/01/>


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[ibc at aliax.net]

2008/11/18 Saurabh Aggarwal <saurabh_aggarwal@hotmail.com>:

[jonas.gauffin at gmail...]

The rfc also describes why:

SIP provides a mechanism to represent common header field names in an
abbreviated form. This may be useful when messages would otherwise
become too large to be carried on the transport available to it
(exceeding the maximum transmission unit (MTU) when using UDP, for
example). These compact forms are defined in Section 20. A compact
form MAY be substituted for the longer form of a header field name at
any time without changing the semantics of the message. A header
field name MAY appear in both long and short forms within the same
message. Implementations MUST accept both the long and short forms
of each header name.


On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo <ibc@aliax.net (ibc@aliax.net)> wrote:

[saurabh_aggarwal at ho...]

Thanks, how do I "enable" this in freeswitch? Can this be done through the SIP configuration file?

-Saurabh



Date: Tue, 18 Nov 2008 12:05:18 +0100
From: jonas.gauffin@gmail.com
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue

The rfc also describes why:

SIP provides a mechanism to represent common header field names in an
abbreviated form. This may be useful when messages would otherwise
become too large to be carried on the transport available to it
(exceeding the maximum transmission unit (MTU) when using UDP, for
example). These compact forms are defined in Section 20. A compact
form MAY be substituted for the longer form of a header field name at
any time without changing the semantics of the message. A header
field name MAY appear in both long and short forms within the same
message. Implementations MUST accept both the long and short forms
of each header name.



On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo <ibc@aliax.net> wrote:

[anthony.minessale at g...]

the only reliable answer is use TCP.

The RFC is daft in this matter.
They say when it's bigger than mtu to automatically use TCP instead.
And timeout for 10 seconds then fall back to UDP.

Its mandatory in SIP to support both TCP and UDP up to 64k per packet.
As you can see, since barely anything will do this right, your best bet is to only use TCP when you have this kind of traffic.


On Tue, Nov 18, 2008 at 5:52 AM, Saurabh Aggarwal <saurabh_aggarwal@hotmail.com (saurabh_aggarwal@hotmail.com)> wrote: