[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection,
SIP Timing Issue?? (Solved)
Aaron Clauson
aza at azaclauson.com
Thu Nov 24 14:04:18 MST 2005
Hi,
I got the person to force the G729 codec on their Linksys WRT54GP2 and
forced it on Asterisk as well. The person then managed to get a single call
out but all subsequent call set ups failed with the same 488 error.
I went back over my SIP traces and noticed that the Cseq's were often out of
order or duplicated. This looked a lot more like the cause and was more
inline with a timing issue which would explain why it was only happening
over satellite. I did some more digging and came across the SIP timing
settings defined in the SIP RFC. I didn't get a chance too read exactly the
mecahnism but one of these settings does seem to be the interval between
resending INVITE requests.
The good news for me and anybody else reading this is with the same problem
is that changing the SIP T1 parameter does get the INVITE requests through.
It's on the SIP configuration page for the Linksys/Sipura devices. In this
case it was changed from the default 0.5s to 2s and then finally to 4s after
which outgoing call set up reliably worked.
In addition it does look like there is a bug in the Asterisk SIP channel
possibly to do with getting confused about receiving a bunch of INIVTE
requests with the same Cseq and stale nonces. It could be related to the
recent 403 problem for the Asterisk SIP channel and the Sipura REGISTER
requests with stale nonces. I will attempt to replicate the SIP dialogue and
produce a SIP trace and if successful file a bug report.
Aaron
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Aaron Clauson
> Sent: 24 November 2005 03:07
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec
> Rejection,SIP Timing Issue??
>
> Hi,
>
> Thanks for the tip I'll try it out. That would explain some
> situations where
> one of the peeople concerned was mucking around with the
> codec settings on
> the PAP2 and managed to get some calls out.
>
> It's a bit baffling how the Linksys devices will get INVITES
> through without
> G.729 being set across non-satellite links and yet can't get
> the very same
> INVITE through across a satellite link. Fair enough if it was
> the Linksys
> generating the 488 during the INVITE negotiation but how does
> Asterisk even
> know the difference??
>
> Aaron
>
> > -----Original Message-----
> > From: Jason p [mailto:voiceoveripguru at gmail.com]
> > Sent: 24 November 2005 02:25
> > To: aza at azaclauson.com; Asterisk Users Mailing List -
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec
> > Rejection, SIP Timing Issue??
> >
> > I had the same problem when we were setting up these boxes
> > after katrina. What i found is that they will only do one
> > G729 session at a time. so that mesg that your showing is
> > that its trying to register two chans as 729. what i did to
> > get around this was to turn off fource prefered codec on one
> > line. This threw me for a loop also but trust me this is the
> > fix, and yes you can only make one 729 call at a time.
> >
> >
> > Jason Price
> >
> >
> > On 11/23/05, Aaron Clauson <aza at azaclauson.com> wrote:
> >
> > Hi,
> >
> > I have a very strange Asterisk SIP call signalling
> > problem that is proving
> > extremely difficult to track down. The problem is that
> > any SIP INVITE
> > request that is coming into Asterisk over a satellite
> > connection from a
> > Linksys Router or PAP2 is getting a "Not Acceptable
> > Here (codec error)" from
> > Asterisk. I've done all the normal checks on the
> > allowed codecs in sip.conf
> > but to no avail.
> >
> > I've even gone as far as writing a basic SIP stack to
> > authenticate and send
> > the INVITE request to Asterisk with exactly the same
> > SDP payload to let me
> > brute force different options in the SDP request to try
> > an narrow it down
> > that way. The preplexing thing from that length
> > exercise is that if exactly
> > the same INVITE request comes in from my app across the
> > same satellite
> > connection to Asterisk it gets 200 Ok'ed but coming
> > from the Linksys PAP2 or
> > WRT54GP2 it gets 488 Codec Not Acceptable Here'ed.
> >
> > The first time this happened we went through all the
> > usual checks and got
> > nowhere and the person drifted off and it was put down
> > to something speicifc
> > to that set up/connection. But now it's cropped up
> > again with a different
> > person who also just happens to be on a satellite
> > connection but from a
> > different provider, although it is possible both
> > providers use the same
> > infrastructure. In both cases incoming calls to the
> > Linksys devices worked
> > correctly it's just the outgoing calls from the devices
> > to Asterisk that are
> > getting the rejection. In the second case we can't put
> > it down to something
> > to do with the connection because the person has a
> > Vonage service working no
> > problems across the same satellite link we are getting
> > the rejection on.
> >
> > The SIP trace is below and I'm wondering if anybody has
> > ever seen something
> > similar. The only thing I can think of is that it's
> > somehow a timing issue I
> > can't see how it can be a codec issue since the exactly
> > the same SDP payload
> > will get OK'ed if coming from my app. Is the Asterisk
> > SIP stack sensitive to
> > the any timings in the INVITE request? It seems highly
> > unlikely but I just
> > can't think of anything else.
> >
> > INVITE sip:018XXX at sip.XXX SIP/2.0
>
> > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
> > From: XXX <sip:XXX at sip.XXX>;tag=831f2cca367c3ddfo1
> > To: <sip:018XXX at sip.xxx>
> > Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> > CSeq: 103 INVITE
> > Max-Forwards: 70
> > Proxy-Authorization: Digest
> >
> > username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:018X
> > XX at sip.XXX",al
> > gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
> > Contact: XXX <sip:XXX at 192.168.1.248:5061>
> > Expires: 240
> > User-Agent: Linksys/PAP2-3.1.3(LS)
> > Content-Length: 424
> > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> > Supported: x-sipura
> > Content-Type: application/sdp
> >
> > v=0
> > o=- 418210 418210 IN IP4 192.168.1.248
> > s=-
> > c=IN IP4 192.168.1.248
> > t=0 0
> > m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:18 G729a/8000
> > a=rtpmap:96 G726-40/8000
> > a=rtpmap:97 G726-24/8000
> > a=rtpmap:98 G726-16/8000
> > a=rtpmap:100 NSE/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=ptime:30
> > a=sendrecv
> >
> >
> > --------------------------------------------------------------
> > --------------
> > ----
> >
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/UDP
> >
> > 192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061
> > From: xxx <sip:XXX at sip.xxx>;tag=831f2cca367c3ddfo1
> > To: <sip:018xxx at sip.xxx>;tag=as17d663fb
> > Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> > CSeq: 103 INVITE
> > User-Agent: asterisk
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> > Contact: <sip:018xxx at xxx>
> > Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3"
> > Content-Length: 0
> >
> >
> >
> > --------------------------------------------------------------
> > --------------
> > ----
> >
> > ACK sip:018xxx at sip.xxx SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b
> > From: xxx < sip:xxx at sip.xxx <mailto:sip:xxx at sip.xxx>
> > >;tag=831f2cca367c3ddfo1
> > To: <sip:018xxx at sip.xxx>;tag=as50c8f92d
> > Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> > <mailto:c71dab66-43f06ff3 at 192.168.1.248>
> > CSeq: 102 ACK
> > Max-Forwards: 70
> > Proxy-Authorization: Digest
> >
> > username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018x
> > xx at sip.xxx ",al
> > gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f"
>
> > Contact: xxx <sip:xxx at 192.168.1.248:5061>
> > User-Agent: Linksys/PAP2-3.1.3(LS)
> > Content-Length: 0
> >
> >
> >
> > --------------------------------------------------------------
> > --------------
> > ----
> >
> > INVITE sip:018xxx at sip.xxx SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
> > From: xxx < sip:xxx at sip.xxx <mailto:sip:xxx at sip.xxx>
> > >;tag=831f2cca367c3ddfo1
> > To: <sip:018xxx at sip.xxx>
> > Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> > CSeq: 103 INVITE
> > Max-Forwards: 70
> > Proxy-Authorization: Digest
> >
> > username="xxx",realm="asterisk",nonce="489bfe04",uri="sip:018x
> > xx at sip.xxx",al
> > gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
> > Contact: xxx <sip:xxx at 192.168.1.248:5061>
> > Expires: 240
> > User-Agent: Linksys/PAP2-3.1.3(LS)
> > Content-Length: 424
> > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> > Supported: x-sipura
> > Content-Type: application/sdp
> >
> > v=0
> > o=- 418210 418210 IN IP4 192.168.1.248
> > s=-
> > c=IN IP4 192.168.1.248
> > t=0 0
> > m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:18 G729a/8000
> > a=rtpmap:96 G726-40/8000
> > a=rtpmap:97 G726-24/8000
> > a=rtpmap:98 G726-16/8000
> > a=rtpmap:100 NSE/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=ptime:30
> > a=sendrecv
> >
> >
> > --------------------------------------------------------------
> > --------------
> > ----
> >
> > SIP/2.0 488 Not Acceptable Here (codec error)
> > Via: SIP/2.0/UDP
> >
> > 192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061
> > From: xxx < sip:xxx at sip.xxx <mailto:sip:xxx at sip.xxx>
> > >;tag=831f2cca367c3ddfo1
> > To: <sip:018xxx at sip.xxx>;tag=as17d663fb
> > Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> > <mailto:c71dab66-43f06ff3 at 192.168.1.248>
> > CSeq: 103 INVITE
> > User-Agent: asterisk
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> > Contact: <sip:018xxx at xxx>
> > Content-Length: 0
> >
> > Thanks,
> >
> > Aaron
> >
> >
> > _______________________________________________
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> >
> >
> >
>
>
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