Thanks Jordan, for having a look at this matter. Yes, that is what Asterisk 11 is sending. Here are complete sip debugs from Asterisk attached. Please refer to IP mapping from OP to have a better understanding.
Is there any way of getting it off from SIP parser on compile time as I am not using this feature and do not intend to use in future. On Wed, Jan 16, 2013 at 7:01 PM, Matthew Jordan <[email protected]> wrote: > On 01/16/2013 07:28 AM, Salman Zafar wrote: > > Hello All, > > I am having a bit peculiar problem with Asterisk 11 for a > > carrier. This carrier shares quite some information in SDP header, which > > should not be the problem, however what happen is as follow: > > > > > > Carrier----> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right > > after answering call drops... Carrier send a BYE with (cause 79: service > > or option not implemented). > > > > *NOTE: Please refer to complete SIP traces attached. * > > * > > * > > *Also Note:* > > _Carrier_: 62.61.147.214 > > _Proxy_: 77.X.X.X:5060 > > _Asterisk11_: 77.X.X.X:5080 > > > > *_Here is Invite SDP from Carrier -> Proxy -> Asterisk 11_* > > > > INVITE sip:[email protected] SIP/2.0 > > v=0 > > o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214 > > s=Phone-Call > > c=IN IP4 77.X.X.X > > t=0 0 > > m=audio 53372 RTP/AVP 8 118 18 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:118 PCMA/8000 > > a=gpmd:118 vbd=yes > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=ptime:20 > > a=sendrecv > > a=rtcp:53373 IN IP4 77.X.X.X > > m=image 56854 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:14400 > > a=T38FaxMaxBuffer:1024 > > a=T38FaxMaxDatagram:122 > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxUdpEC:t38UDPRedundancy > > > > /*_SDP:After Answered by Asterisk 11_*/ > > v=0 > > o=root 164966782 164966782 IN IP4 77.X.X.X > > s=Asterisk v11.0.1 > > c=IN IP4 77.X.X.X > > t=0 0 > > m=audio 12636 RTP/AVP 18 8 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:8 PCMA/8000 > > a=ptime:20 > > a=sendrecv > > *_m=image 0 udptl t38_* > > > The appropriate way for Asterisk to indicate that it does not support a > media stream is to set the port number to 0. We have to inform the > offerer that we don't support the media stream; removing it from the SDP > completely is not allowed. > > Per RFC 3264, section 6: > > " An offered stream MAY be rejected in the answer, for any reason. If > a stream is rejected, the offerer and answerer MUST NOT generate > media (or RTCP packets) for that stream. To reject an offered > stream, the port number in the corresponding stream in the answer > MUST be set to zero. " > > > I have tired by disabling/unloading fax modules as *I am not using* them > > but no results. Secondly, also tried tweaking of udptl ever-odd nothing > > worked. > > You've configured your system to not support fax correctly. Asterisk is > rejecting the offered image stream accordingly. > > > The same carrier works for Asterisk 1.6.X and the only difference I have > > notice so far is the above underlined line in Answered SDP -> m=image 0 > > udptl t38. I think if I some how do not advertise udptl here i would be > > able to avoid this scenario. I have tried multiple ways to strip off SDP > > from incoming INVITE at SIP Proxy level but it is not SDP wise enough. > > > > I'm not sure what 1.6.x is sending. It's possible that it just > completely removed the stream from the SDP answer, which is wrong. > > Section 6 again: > > "For each "m=" line in the offer, there MUST be a corresponding "m=" > line in the answer." > > > *Note:* > > > > In Asterisk 1.6 => WARNING[32671]: chan_sip.c:8833 process_sdp: > > Unsupported SDP media type in offer: image 59978 udptl t38 > > In Asterisk 11 => WARNING[18748][C-0000002f]: chan_sip.c:10277 > > process_sdp: Failed to initialize UDPTL, declining image stream > > > > > > An initial glance at this makes me think your carrier is doing something > wrong. Just to check, however, is the SDP answer you pasted the entire > SDP that Asterisk 11 responds with? Specifically, are there no format > attributes for the image stream in the SDP that Asterisk responds with? > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Regards ************************** Muhammad Salman ***************************
; *********************************************** INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:[email protected];lr;ftag=1c1638837366;did=4c1.c49b9bd6> Via: SIP/2.0/UDP 77.X.X.X;branch=z9hG4bKbf17.c3dd9193.0 INVITE from Carrier to Sip Proxy -> Asterisk 11 Via: SIP/2.0/UDP 62.61.147.214;rport=5060;received=62.61.147.214;branch=z9hG4bKac1638847684 Max-Forwards: 69 From: <sip:[email protected]>;tag=1c1638837366 To: <sip:[email protected]> P-CallKey: [email protected] Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]> Supported: em,100rel,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.026.007 Privacy: none P-Asserted-Identity: <sip:[email protected]> Content-Type: application/sdp Content-Disposition: session Content-Length: 493 P-hint: outbound v=0 o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214 s=Phone-Call c=IN IP4 77.X.X.X t=0 0 m=audio 53372 RTP/AVP 8 118 18 a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtcp:53373 IN IP4 77.X.X.X m=image 56854 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy ; *********************************************** Reply-> Answer() from Asterisk 11 -> SIP Proxy -> Carrier [2013-01-16 12:36:53] -- Executing [69609000@origination_incoming:1] Answer("SIP/RTSIP-In-00000025", "") in new stack [2013-01-16 12:36:53] Audio is at 12636 [2013-01-16 12:36:53] Adding codec 100008 (g729) to SDP [2013-01-16 12:36:53] Adding codec 100004 (alaw) to SDP [2013-01-16 12:36:53] <--- Reliably Transmitting (no NAT) to 77.X.X.X:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 77.X.X.X;branch=z9hG4bK8b41.01141c16.0;received=77.X.X.X Via: SIP/2.0/UDP 62.61.147.214;rport=5060;received=62.61.147.214;branch=z9hG4bKac1486322350 Record-Route: <sip:[email protected];lr;ftag=1c1486312034;did=184.10763da5> From: <sip:[email protected]>;tag=1c1486312034 To: <sip:[email protected]>;tag=as77bfa472 Call-ID: [email protected] CSeq: 1 INVITE Server: Asterisk v11.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: <sip:[email protected]:5080> Content-Type: application/sdp Content-Length: 242 v=0 o=root 164966782 164966782 IN IP4 77.X.X.X s=Asterisk v11.0.1 c=IN IP4 77.X.X.X t=0 0 m=audio 12636 RTP/AVP 18 8 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv m=image 0 udptl t38 <------------> [2013-01-16 12:36:53] <--- SIP read from UDP:77.X.X.X:5060 ---> ACK sip:[email protected]:5080 SIP/2.0 Via: SIP/2.0/UDP 77.X.X.X;branch=z9hG4bK8b41.01141c16.2 Via: SIP/2.0/UDP 62.61.147.214;rport=5060;received=62.61.147.214;branch=z9hG4bKac1487533482 Max-Forwards: 69 From: <sip:[email protected]>;tag=1c1486312034 To: <sip:[email protected]>;tag=as77bfa472 Call-ID: [email protected] CSeq: 1 ACK Contact: <sip:[email protected]> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.026.007 Content-Length: 0 <-------------> Righ Away BYE sent from carrier to Proxy to Asterisk11 [2013-01-16 12:36:53] --- (13 headers 0 lines) --- [2013-01-16 12:36:53] <--- SIP read from UDP:77.X.X.X:5060 ---> BYE sip:[email protected]:5080 SIP/2.0 Via: SIP/2.0/UDP 77.X.X.X;branch=z9hG4bK5b41.aca53c42.0 Via: SIP/2.0/UDP 62.61.147.214;rport=5060;received=62.61.147.214;branch=z9hG4bKac1487558712 Max-Forwards: 69 From: <sip:[email protected]>;tag=1c1486312034 To: <sip:[email protected]>;tag=as77bfa472 Call-ID: [email protected] CSeq: 2 BYE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.026.007 Reason: Q.850 ;cause=79 Content-Length: 0
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
