[asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop

Thu Jan 17 00:15:32 CST 2013

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 <mjordan at digium.com> 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:69609000 at 77.X.X.X 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 --
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-- 
Regards

**************************
Muhammad Salman
***************************
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; *********************************************** INVITE sip:69609000 at 77.X.X.X SIP/2.0

Record-Route: <sip:69609000 at 77.X.X.X;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:923366132989 at ag07-hor.aplus.dk>;tag=1c1638837366
To: <sip:69609000 at 77.X.X.X>
P-CallKey: 1638836614161201314220 at 62.61.147.214
Call-ID: 1638836614161201314220 at 62.61.147.214
CSeq: 1 INVITE
Contact: <sip:923366132989 at 62.61.147.214>
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:923366132989 at ag07-hor.aplus.dk>
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 at 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:69609000 at 77.X.X.X;lr;ftag=1c1486312034;did=184.10763da5>
From: <sip:923366132989 at ag07-hor.aplus.dk>;tag=1c1486312034
To: <sip:69609000 at 77.X.X.X>;tag=as77bfa472
Call-ID: 14863112831612013143653 at 62.61.147.214
CSeq: 1 INVITE
Server: Asterisk v11.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:69609000 at 77.X.X.X: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:69609000 at 77.X.X.X: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:923366132989 at ag07-hor.aplus.dk>;tag=1c1486312034
To: <sip:69609000 at 77.X.X.X>;tag=as77bfa472
Call-ID: 14863112831612013143653 at 62.61.147.214
CSeq: 1 ACK
Contact: <sip:923366132989 at 62.61.147.214>
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:69609000 at 77.X.X.X: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:923366132989 at ag07-hor.aplus.dk>;tag=1c1486312034
To: <sip:69609000 at 77.X.X.X>;tag=as77bfa472
Call-ID: 14863112831612013143653 at 62.61.147.214
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