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Cisco Cube & T.38 Fax protocol - Call Manager 10.5

Yannick Vranckx
Level 2
Level 2

Hello,

I wish some assistance in finding out an issue: Recently we upgraded a customers call manager from 8.6 to 10.5, the upgrade went smooth. There is 1 issue with the T.38 Fax protocol, we have a Faxination server which handles the faxes to e-mail etc. This is configured towards the call manager, so for the fax number we have a route pattern which is forwarded to a SIP connection that is connected to the Faxination server.

 

So for example 9999 is a fax number that has a route pattern towards a SIP Trunk to the Faxination server.

 

This doesn't work anymore on the Call Manager 10.5, the voicegateways are configured as CUBE which have a sip connection to the provider.

I believe the problem is situated on the Cisco Cube and not on the Call Manager: I have traced in wireshark from Call Manager and also from the fax server, there is a T.38 max buffer set of 14000. For a weird reason when the T.38 transmission passes the Cisco CUBE the T.38 max buffer is set towards 200. At first i thought it was the provider but it isn't, it's my own cisco cube that sends it towards the provider causing the fax call to fail.

I can't even call inwards to the fax number; it doesn't dial, it automatically hangs up with a small beep.

 

 

For example a trace from the Cisco Cube:

***************************

Received:
INVITE sip:XXXXXXX@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKac03120ad68f
From: <sip:XXXXXXX@X.X.X.X>;tag=43717~cac1a8e8-91f3-4735-bc48-b68717f25816-46700953
To: <sip:XXXXXXXX@X.X.X.X>;tag=D0EBE408-2387
Date: Mon, 25 Aug 2014 12:47:31 GMT
Call-ID: CF684087-2B8C11E4-B2B0AF09-C66FE7A3@X.X.X.X
Supported: timer,resource-priority,replaces
Cisco-Guid: 3479636023-0730599908-2997530377-3329222563
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence, kpml
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
Remote-Party-ID: <sip:XXXX@X.X.X.X>;party=calling;screen=no;privacy=off
Contact: <sip:XXXX@X.X.X:XXXX>
Content-Type: application/sdp
Content-Length: 360

v=0
o=CiscoSystemsCCM-SIP 43717 2 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=image 4012 udptl t38
a=T38FaxVersion:1
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxMaxBuffer:14000
a=T38FaxMaxDatagram:1500

********************************************

I have bolded the important parts; This is a trace from the Call Manager to the Cube VoiceGateway, notice that the T38FaxMaxBuffer is set to: 14000

 

If i continue the trace, there will be a next trace that will send the information from the local Cisco Cube towards the provider:

***********************************************

Sent:
INVITE sip:XXXXXXXX@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK3A902266B
From: <sip:XXXXXXX@X.X.X.X>;tag=D0EBE41C-E5B
To: <sip:XXXXXXXX@X.X.X.X;pstn-params=808382808883>;tag=gK0219eb8c
Date: Mon, 25 Aug 2014 12:47:31 GMT
Call-ID: 687997967_7844794@X.X.X.X
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3479636023-0730599908-2997530377-3329222563
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1408970851
Contact: <sip:XXXXXXX@X.X.X.X:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 326

v=0
o=CiscoSystemsSIP-GW-UserAgent 347 1499 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=image 26298 udptl t38
c=IN IP4 X.X.X.X
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

**************************************************************

Here we can see that the Fax Max Buffer has been changed somehow to a value of 200, it came from 14000. So i'm thinking it's something going wrong here. In the wireshark traces i've pulled from Call Manager to the Fax server i can see the perfect SIP setup but ofcourse no RTP stream because it just doesn't happen.

 

I've been thinking to upgrade the IOS of the router, could that help? Or is this a hidden setting on the Cube that i can adjust. Googling didn't bring me a clear answer.

 

 

Kr,

 

 

Yannick Vranckx

19 Replies 19

Manish,

 

The issue is resolved, the problem wasn't the negotiation of the version , it was the max buffer and datagram sent out rom the CUBE to the faxination server.

The FOIP device rejects these settings because the buffer is smaller than the datagram. We needed to modify them so the buffer is larger then the datagram.

 

We had below configuration on the SIP voice gateways

voice class sip-profiles 1
response ANY sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
request ANY sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
request ANY sdp-header Attribute modify "a=T38FaxMaxDatagram:320" "a=T38FaxMaxDatagram:180"
response ANY sdp-header Attribute modify "a=T38FaxMaxDatagram:320" "a=T38FaxMaxDatagram:180" 

 

This profile we activated on all the voip dial peers to the Call Manager and that did the trick

 

 

Kind Regards,

 

 

Yannick Vranckx

 

 

 

Hi Yannick,

 

I think I might be running into a similar issue as you with the buffer size at 200 and max datagram at 320. I guess this might be a SIP default from the CUBE. 

How did you come up with the value of 180 for your max datagram? or it is because it is smaller than 200?  I am planning on trying these setting to see if it fixes my faxing issues.

 

Regards,

Allan

Dear,

 

We had to actually add a sip profile to get this sorted on the gateway. I don't know the exact syntax no more, i'll look it up asap and paste it here. 

Otherwise it would have never worked.

 

Kr,

 

We are using this approach for similar issues.

Syntax to add sip profile to dial-peer is:

dial-peer voice xx voip
sip-profiles XXX
sip-profiles XXX inbound

It's looking like the T38 is sticking at version 0 (or 1 which is odd.)   I'm looking at line 419 of the debugging output and it's saying it's negotiated at version 1.   When searching through the rest of the output I see T38FaxVersion is staying at 0.