DTMF Problem in SIP Trunk

payamkhosravi — Mon, 18 Mar 2019 13:18:54 GMT

I have a problem in case of detecting DTMF

We have got a SIP Trunk from ITSP and everything is ok except DTMF.

The sip trunk is between ITSP and Cisco 3945 Router

ITSP <-> 3945 <-> CUCM 10.5

I test all of method such as rtp-nte , h245-alphanumeric and h245-signal ,sip-info,sip-kpml ,.... in dial-peer toward itsp

ITSP say that he send dtmf with RFC2833 standard (it equal to rtp-net as i know) but when i get "debug ccsip message" , the results shows the dtmf with rfc2833 does not send to us

Aug 16 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:42584000@10.198.5.174:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.105.40.34:5060;branch=z9hG4bKejhh8jixkobhnb7ykvi87vuj8

Call-ID: SBCnhcthc33gnsw3ujt5tssgtuivsmmjhtc@SoftX3000

From: <sip:9123008963@10.105.40.34>;tag=c3cx3vcw-CC-40

To: <sip:42584000@10.198.5.174;user=phone>

CSeq: 1 INVITE

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER

Max-Forwards: 69

Supported: 100rel,timer

User-Agent: Huawei SoftX3000 V300R010

Session-Expires: 300

Min-SE: 90

Contact: <sip:9123008963@10.105.40.34:5060;user=phone>

Content-Length: 374

Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34

s=Sip Call

c=IN IP4 10.105.40.34

t=0 0

m=audio 27762 RTP/AVP 8 0 18 4 2 98 99 102

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:98 G726-40/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:102 G726-24/8000

a=ptime:20

a=fmtp:18 annexb=no

This is a invite message (with sdp) from ITSP

As you see the line with red color must have a code with number of 101 but instead have code with number of 18

In my "debug ccsip media" output show that the method of negotiation between me and itsp is "inbound voice"

this is my router config:

voice service voip
no ip address trusted authenticate
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface FastEthernet0/0/1
bind media source-interface FastEthernet0/0/1
min-se 300 session-expires 300
!

dial-peer voice 2 voip --------------> from Router to CUCM and vice versa
translation-profile outgoing toos
destination-pattern 42584...
session protocol sipv2
session target ipv4:10.20.30.70
codec g711ulaw
dtmf-relay rtp-nte
!
dial-peer voice 10 voip ------------------> from Router to ITSP and vice versa
destination-pattern .T
session protocol sipv2
session target ipv4:10.105.40.34
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw

I configured cucm with a sip trunk toward my router with MTP enabled and RFC2833 preferred

BUT THERE IS NO DTMF DETECTION IN INBOUND AND OUTBOUND CALLS

I even test dial-peer 10 without any dtmf-relay method because I wanna configure it with inbound voice method but it does not work

I change the codec but does not solve the problem

There is a interesting point and that is , if use Elastix instead of 3945 router and configure dtmf method between elastix and itsp as "inbound" and dtmf method between elastix and cucm as "rfc2833" everything is OK ( ITSP<--Inbound--> Elastix <--rfc2833--> CUCM)

Please give me a solution to solve the problem between Cisco 3945 and ITSP

Regards

Hi

payamkhosravi — Sun, 28 Aug 2016 18:46:47 GMT

The problem was solved by using hardware transcoding for DTMF interworking

It would be more accurate to

Jonathan Schulenberg — Sun, 28 Aug 2016 21:49:42 GMT

It would be more accurate to say that the problem was worked around by using a transcoding resource. The correct resolution here would be to open a trouble ticket with the ITSP. I agree that the INVITE message you show does not include RFC2833 advertisement in the SDP offer. That is a problem that only the carrier can rectify.

hello Payamkhosravi2000, I

ev1205 — Thu, 29 Jun 2017 15:13:02 GMT

hello Payamkhosravi2000, I have similar problem like you but only with one international client, my other international calls to IVRs do not have problem with DTMF.

I have transcoder for my calls.

Could you please explain me how was your solution?

Best regards

topic It would be more accurate to in TelePresence and Video Infrastructure

DTMF Problem in SIP Trunk

Hi

It would be more accurate to

hello Payamkhosravi2000, I