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DTMF problem with  Cisco 7300 VOIP gateway

Posted on 2010-09-01
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Last Modified: 2012-05-10
I am taking in a call from a VOIP customer through a Cisco 7300 and terminating this out to a Tier 1 carrier.  It appears the the DTMF tones are being distorted or not recognized and the problem is arising in my Cisco.  I have heard that it is recommended that I use g711, unfortunately this is not possible.  I have also ensured that the command:  dtmf-relay h245-alphanumeric is being used in the corresponding dial-peer.  Can anyone help me with this problem?  Any input is gladly appreciated.

  -Chris
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Question by:shadeytrees
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7 Comments
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33585355
Hi Chris,
With the limited information provided, it is not possible to provide a solution.
Can you provide details about your network topology end-to-end? Please describe the voip endpoints between the caller and called parties, and the the type of signalling types used along the path. Then provide your router's voip configuration, including transcoders etc.
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Author Comment

by:shadeytrees
ID: 33598319
Thanks for your attention Abahar,

This is an Internet VOIP connection using SIPv2 the customer has a Sonus and is sending DTMF tones to my cisco 7300 (C7301-ADVENTERPRISEK9-M), Version 12.4(11)T4 we then pass the call onto a Sansay VSX-S2-2U integrated session switch which then is connected to the local PTT though a cisco 5300.  The calls seem to work fine when I test from a Cisco 2600>Cisco 7300>Sansay>Cisco 5300>PTT.  However calls from the Sonus to my network do not pass DTMF, but they do through the second providers network.  Any thoughts?
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LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 100 total points
ID: 33601664
Apparently there are some RTP timestamp issues communicating with Sonus gateways.
On Cisco I have found The following explanantion. But this bug CSCtd68173 seems to be for the opposite direction. Cisco DTMF is not recognized by Sonus.
Timestamp not accurate in the DTMF sent by CME Symptoms: Outbound DTMF may fail intermittently over a SIP Trunk from Cisco
UC 500.

Conditions: This symptom is observed when the following conditions are
present:
- Using Cisco IOS Release 12.4.22YB4 or Release 15.0.1XA on Cisco UC 500
- SIP trunk uses RFC2833 for DTMF
- Call is outbound from Cisco IP Phone to PSTN over SIP Trunk
- SIP Trunk provider gateway is Sonus GXS (v6.4).

Workaround:
- Use Cisco IOS Release 12.4(11)XW10 on the Cisco UC 500 if possible; or,
- SIP trunk provider Sonus GXS gateway should be upgraded to v6.5.5 or higher.
 
 
I had found this explanation on http://wiki.freeswitch.org/wiki/RTP_Issues there is an Asterisk patch for Sonus as well.
Sonus wrongly expects that, when sending a multi-packet 2833 DTMF event, the sender should
increment the RTP timestamp in each packet when, in reality, the sender should send the
same exact timestamp and increment the duration field in the 2833 payload. This allows a
reconstruction of the duration if any of the packets are lost.

final_duration - initial_timestamp = total_samples

However, if the duration value exceeds the space allocated (16 bits), the sender should
increment the timestamp one unit and reset the duration to 0.

Always sending a duration of 0 with a new timestamp should be tolerated but is rarely
intentional and is mistakenly done by many devices. The issue is that the Sonus expects
everyone to do it this way instead of tolerating either way. Sonus will actually ignore
every packet with the same timestamp before concluding if it's DTMF.
 
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Author Comment

by:shadeytrees
ID: 33612242
Abahar,

  I appreciate your insight into this.  My customer with the Sonus gateway is investigating this now.  I will advise upon our correspondence.  Thank you again for you assistance this is very helpful.

-Chris
0
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33612347
Hi Chris,
I hope it is caused by a software issue and upgrading the software will get it working. Fingers crossed.
Regards,
Alex
0
 

Accepted Solution

by:
shadeytrees earned 0 total points
ID: 33612471
I found the answer.  I just added  "rtp-nte sip-notify" to my "dtmf-relay  h245-signal  h245-alphanumeric"  string in the customers dial-peer and DTMF passed without fault.  teh command looks as follows:

dial-peer voice 1 voip
dtmf-relay rtp-nte h245-signal sip-notify h245-alphanumeric
!

Router(config-dial-peer)# dtmf-relay rtp-nte
Allows DTMF relay using NTE RTP packets. DTMF tones are encoded in the NTE format and transported in the same RTP channel as the voice.

  Thanks all for your tips.

 -Chris
 
0
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33614302
Hi Chris,
Thanks for sharing the solution. I am sure other people will find that useful as well.
Regards,
Alex
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