On a related note, my parents complained about dropped SIP calls on a Linksys ATA. Replacing the 3 yr old Aztech DSL modem, and the verizon actiontec router, with a new d-link modem/router has resulted in no dropped calls. So.. buggy modem/routers affecting calls might be more common than you think. -Avi Marcus On Thu, Nov 24, 2011 at 5:34 PM, Charlie Orford < charlie.orford@attackplan.net> wrote:
Hi Chris
I don't know why but replacing the router (a 5 year old Draytek Vigor 2600) with a linux machine + pppoe modem fixed the problem.
No change was needed with the phones and they are still behind NAT (but this time its managed by iptables).
Buggy router I guess....
Hmm it doesn't look like it is a fragmentation problem. Same "400 Bad Request" response even with the INVITE at 995 bytes:
http://pastebin.freeswitch.org/17868 http://pastebin.freeswitch.org/17867
FYI, issuing "ping -f -l 1470 my.fs.server.com" from a windows workstation on the same network as the phone goes through ok (so I think this means as long as the INVITE packet is 1470 bytes or less, it should not get fragmented).
Is there anything else I can debug to track this down?
Charlie
On 24/11/2011 14:59, chrisbware@interfree.it wrote:
Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To test if this is the problema, disable all codecs leaving only "basic codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet should be smaller: if the problem is fixed in this way, packet fragmentation is the cause.
-----Messaggio originale----- Da: Charlie Orford<charlie.orford@attackplan.net> Inviato il: 24 Nov 2011 - 13:47 A: FreeSWITCH Users Help<freeswitch-users@lists.freeswitch.org>
Hi Chris,
I've increased the debugging and taken a new trace:
http://pastebin.freeswitch.org/17864
And here is the output of tport_sip.log:
http://pastebin.freeswitch.org/17865
A person on the #freeswitch IRC channel suggested the problem might be due to packet fragmentation?
Charlie
On 24/11/2011 14:29, chrisbware@interfree.it wrote:
I suspect problem is on SDP but we can't see it on pastebin. -----Messaggio originale----- Da: Charlie Orford Inviato il: 24 Nov 2011 - 10:17 A: FreeSWITCH Users Help
Hi list
I have done some googling but cannot find much info about what causes FS to send a SIP/2.0 400 Bad Request response.
My phones are able to register but when I try a test call (e.g. to
On 24/11/2011 15:27, Charlie Orford wrote: 9196)
FS responds to the INVITE with Bad Request. For a short transcript of this in action see: http://pastebin.freeswitch.org/17863
My setup: I have a clean git head install (cloned yesterday) and am running the unmodified "out of the box" configuration (except I have enabled NDLB-force-rport on the internal profile and set a default poassword). FS is on a public server (no nat) and phones (aastra 57i models) are all natted behind an ADSL router.
Thanks& Regards, Charlie
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