[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

Aaron Clauson aza at azaclauson.com
Wed Nov 23 18:41:07 MST 2005


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:018XXX 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>;tag=831f2cca367c3ddfo1
To: <sip:018xxx at sip.xxx>;tag=as50c8f92d
Call-ID: c71dab66-43f06ff3 at 192.168.1.248
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018xxx 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>;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:018xxx 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>;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>
Content-Length: 0

Thanks,

Aaron





More information about the asterisk-users mailing list