01-04-2014 01:19 PM - edited 03-16-2019 09:06 PM
Hello,
I was wondering the call process between two cisco voip phones with dtmf and rtp packet. For example if caller A dials first then does rtp send a dtmf info packet first and then when caller B picks up then caller A send another rtp voice packet such as hello to caller b. How does dtmf work with rtp as well as voice between a voip call when it initiates and then ends after hang up?
01-05-2014 07:21 AM
In Cisco VoIP telephony, dtmf digits are either passed inband or out of band. In inband voice using rtp-nte, dtmf digits are passed in rtp sreams but uses a different payload type. In out of band method, dtmf are sent in the call signalling and not in the rtp stream...
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-05-2014 10:40 AM
Hello,
I have question regarding "dtmf are sent in the call signalling and not in the rtp stream...". What is difference from call signaling and rtp stream, could I get an example?
Thanks,
01-05-2014 11:55 AM
Horacio,
Call Signaling is the communication path used to set up a new call and to manage/re-negotiate an existing call. The mechanisms involved depend on which protcol stack you are dealing with. The parties involved in the Call Signaling stream also depends on whether you are using a peer-to-peer protocol (like H323) or a client-server protocol (like SCCP, MGCP). Note that H323 and SIP solutions can also be implemented in a client-server architecture.
In the scenario you started with, I am guessing that you have a client-server architecture and are either using SCCP, SIP, or both. This means that an out-of-band (OOB) signaling for DTMF are sent from the station receiving the user's key presses and the CUCM. The CUCM is then relaying that to the receiving party (phone, voice mail, call center, whatever). In some scenarios, you may need to have DTMF cross a border element or voice gateway. This is where all of that DTMF relay lingo comes into play. Basically, just like any other gateway function, the device acting as the arbitration point between two networks needs to be able to ensure requests from one party are properly translated so the next-hop party (or final destination party) can understand what is being requested.
With in-band DTMF signaling, you are sending the frequency tones in the RTP stream and this stream is flowing directly between endpoints (for on-net call scenarios) or flowing through intermediary media hosts if you are using a MTP, transcoder, voice gateway, CUBE, etc. The point is that the call processing agent is not involved with in-band DTMF signaling. Also, you are relying on the fact that the DTMF tones are accurately preserved through the entire media path. This can become a problem if you are doing something like transcoding to G729 somwhere in the media path.
In general, the preference is to leverage out of band DTMF signaling as it is the most reliable method. Of course, you need to ensure that all parties involved in the call understand the OOB method you are applying.
You can take a look at:
http://www.ietf.org/rfc/rfc2833.txt
and if you want to see a sample of a SIP call using RFC2833 for DTMF, check:
http://wiki.wireshark.org/SampleCaptures
Look under "SIP and RTP"
HTH
-Bill
(b) http://ucguerrilla.com
(t) @ucguerrilla
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01-05-2014 03:37 PM
Hello,
In dtmf in band a phone call does not go thru call manager to get routed does that mean it goes thru a h.323 voice gateway instead. How does the process work When a caller from a voip phone dials to another voip phone using dtmf in band. In other words dtmf would indicate who to call however what device uses dftm in band rtp packet to know where to route those voice rtp packets?
01-06-2014 02:18 AM
You are mixing things up.
DTMF and Voice calls are two different things. You cant not have dtmf tones without first of all establishing a voice call.
Phone A calls phone B, CUCM is used for signalling and RTP are sent directly between the two phones. Phone A and Phone B is now connected. Next Phone A or Phone B can then send DTMF tones if either needs to..
It is at this point that DTMF method comes into effect. You can transport those tones using either Inband or OOB (out of band)..
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-06-2014 08:47 AM
Hi Aokanlawon,
In the Analog world when you dialed then dtfm would send tones frequencies which the phone company would know where to route the call. In the VOIP world how is dtfm different when you mean tones are just referring to to the sound the caller hears when he dials but here dtfm tones frequency are not used to route call by call manager. Pretty much what is the different from DTFM in analog world to dtfm in VOIP world usind call manager with dtfm inband and OOB?
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