Aaron Clauson
2005-Nov-23 18:41 UTC
[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??
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@sip.XXX SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f From: XXX <sip:XXX@sip.XXX>;tag=831f2cca367c3ddfo1 To: <sip:018XXX@sip.xxx> Call-ID: c71dab66-43f06ff3@192.168.1.248 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:018XXX@sip.XXX",al gorithm=MD5,response="22f566e03a225047469d73bec5ab640c" Contact: XXX <sip:XXX@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@sip.xxx>;tag=831f2cca367c3ddfo1 To: <sip:018xxx@sip.xxx>;tag=as17d663fb Call-ID: c71dab66-43f06ff3@192.168.1.248 CSeq: 103 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:018xxx@xxx> Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3" Content-Length: 0 ---------------------------------------------------------------------------- ---- ACK sip:018xxx@sip.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b From: xxx <sip:xxx@sip.xxx>;tag=831f2cca367c3ddfo1 To: <sip:018xxx@sip.xxx>;tag=as50c8f92d Call-ID: c71dab66-43f06ff3@192.168.1.248 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018xxx@sip.xxx",al gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f" Contact: xxx <sip:xxx@192.168.1.248:5061> User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 ---------------------------------------------------------------------------- ---- INVITE sip:018xxx@sip.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f From: xxx <sip:xxx@sip.xxx>;tag=831f2cca367c3ddfo1 To: <sip:018xxx@sip.xxx> Call-ID: c71dab66-43f06ff3@192.168.1.248 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="xxx",realm="asterisk",nonce="489bfe04",uri="sip:018xxx@sip.xxx",al gorithm=MD5,response="22f566e03a225047469d73bec5ab640c" Contact: xxx <sip:xxx@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@sip.xxx>;tag=831f2cca367c3ddfo1 To: <sip:018xxx@sip.xxx>;tag=as17d663fb Call-ID: c71dab66-43f06ff3@192.168.1.248 CSeq: 103 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:018xxx@xxx> Content-Length: 0 Thanks, Aaron
Kevin P. Fleming
2005-Nov-23 19:23 UTC
[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??
Aaron Clauson wrote:> 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/8000I don't know what this (NSE) is, but Asterisk certainly doesn't support it. The only way we can debug this is by getting a complete 'sip debug' and 'set verbose' console trace; read the bug posting guidelines at bugs.digium.com and open a bug there with the required information.
Jason p
2005-Nov-23 19:24 UTC
[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@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@sip.XXX SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f > From: XXX <sip:XXX@sip.XXX>;tag=831f2cca367c3ddfo1 > To: <sip:018XXX@sip.xxx> > Call-ID: c71dab66-43f06ff3@192.168.1.248 > CSeq: 103 INVITE > Max-Forwards: 70 > Proxy-Authorization: Digest > username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:018XXX@sip.XXX > ",al > gorithm=MD5,response="22f566e03a225047469d73bec5ab640c" > Contact: XXX <sip:XXX@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@sip.xxx>;tag=831f2cca367c3ddfo1 > To: <sip:018xxx@sip.xxx>;tag=as17d663fb > Call-ID: c71dab66-43f06ff3@192.168.1.248 > CSeq: 103 INVITE > User-Agent: asterisk > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:018xxx@xxx> > Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3" > Content-Length: 0 > > > > ---------------------------------------------------------------------------- > ---- > > ACK sip:018xxx@sip.xxx SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b > From: xxx <sip:xxx@sip.xxx>;tag=831f2cca367c3ddfo1 > To: <sip:018xxx@sip.xxx>;tag=as50c8f92d > Call-ID: c71dab66-43f06ff3@192.168.1.248 > CSeq: 102 ACK > Max-Forwards: 70 > Proxy-Authorization: Digest > username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018xxx@sip.xxx > ",al > gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f" > Contact: xxx <sip:xxx@192.168.1.248:5061> > User-Agent: Linksys/PAP2-3.1.3(LS) > Content-Length: 0 > > > > ---------------------------------------------------------------------------- > ---- > > INVITE sip:018xxx@sip.xxx SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f > From: xxx <sip:xxx@sip.xxx>;tag=831f2cca367c3ddfo1 > To: <sip:018xxx@sip.xxx> > Call-ID: c71dab66-43f06ff3@192.168.1.248 > CSeq: 103 INVITE > Max-Forwards: 70 > Proxy-Authorization: Digest > username="xxx",realm="asterisk",nonce="489bfe04",uri="sip:018xxx@sip.xxx > ",al > gorithm=MD5,response="22f566e03a225047469d73bec5ab640c" > Contact: xxx <sip:xxx@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@sip.xxx>;tag=831f2cca367c3ddfo1 > To: <sip:018xxx@sip.xxx>;tag=as17d663fb > Call-ID: c71dab66-43f06ff3@192.168.1.248 > CSeq: 103 INVITE > User-Agent: asterisk > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:018xxx@xxx> > Content-Length: 0 > > Thanks, > > Aaron > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051123/f2ce57ab/attachment.htm