[asterisk-users] SIP peer / Maximum retries exceeded on transmission
chris at cgb1911.mine.nu
chris at cgb1911.mine.nu
Thu May 3 17:42:13 MST 2007
Hi Everyone,
I was hoping someone might know why I am experiencing a problem with
Asterisk logging the event:
[May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be at sip.externalprovider.com for seqno 669371069 (Critical Response)
This is happening after:
- call is setup, 2 way audio
- call can function correctly for up to 5 minutes, with the external
provider re-inviting every 1 minute
When the problem happens
- external peer re-invites asterisk
- asterisk sends 200 OK
- external peer sends ACK
- asterisk retransmits 200 OK
- external peer sends ack
- ..
- asterisk retransmits 200 OK (Retransmitting #6)
- external peer sends ack
- Asterisk logs the above message about maximum retries exceeded,
and sends BYE to the inside SIP UA.
The network configuration is as follows:
phone <--> alternative SIP server <--> Asterisk <-NAT-> External peer
The alternative SIP server is not a B2BUA, just SIP proxy. Now,
sometimes a call can work without any problems, but not as often as
when the above symptoms are experienced.
The references I've found online about this type of problem suggest
NAT as being the culprit, but in this case, Asterisk is logging it's
reception of the ACK but deciding to ignore it and retransmit the
200 OK anyhow. I'm guessing in other cases people suspect is' NAT
because they believe SIP isn't getting back trhough after a period of
time.
I was using 1.4.2, but found this changelog today for 1.4.3:
ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3
2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul at odt.east.telecom.kz>
* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
sends ACK not on OK message only (when remote party answers) but
on RINGING message too, so when we send 200 OK message, we get
unidentified ACK message (because INVITE acknowledged on RINGING
message already), so 200 OK retransmits within its retransmission
interval then call gets dropped. If someone else knows how to
provide workaround for such cases, please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
I've upgraded to 1.4.4 but the problem still persists. The above
changelog doesn't sound exactly like what I"m experiencing but maybe
it's related.
Attached is my sip.conf, extensions.conf, and (debug = 10) logs for
one example. I don't know what else might be needed to help anyone
assist me in this problem - let me know if I missed something.
It *feels* like an Asterisk bug but maybe a SIP expert can spot the
problem in signalling/RFC conformance..
Thanks in advance,
Chris Bennett
-------------- next part --------------
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=proxy.myhostname
disallow=all
allow=alaw
sipdebug = yes
recordhistory=yes
dumphistory=yes
register => <authstuff>@sip.externalpeer.com
externhost=proxy.myhostname
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
nat=never
canreinvite=no
[authentication]
auth = <authstuff>@sip.externalpeer.com
[provider]
type=peer
username=<myusername>
secret=<mysecret>
fromuser=<myusername>
fromdomain=sip.externalpeer.com
host=sip.externalpeer.com
nat=never
canreinvite=no
[1111]
type=friend
username=1111
secret=<secret>
host=dynamic
context=tutorial
nat=never
insecure=invite
qualify=yes
-------------- next part --------------
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
[tutorial]
exten => _XXXXXXX.,1,Dial(SIP/${EXTEN}@provider,,r)
-------------- next part --------------
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