[SIPForum-discussion] Regarding Sending DTMF over RTP

Manoj Priyankara manoj0915 at gmail.com
Tue Aug 30 04:35:13 UTC 2011


Hi,

Yes. you can have two payload types in a single call/session.

BR,
Manoj

On Mon, Aug 29, 2011 at 3:07 AM, Shrivastava, Keshav (Keshav) <
kshrivastava at avaya.com> wrote:

>  *Hi,*
>
> * *
>
> *This is one of the old Questions that I am trying to search an answer.
> Please help. *
>
> * *
>
> *I'm trying to nail down my understanding of how the RFC 2833 DTMF
> Telephone-event RTP Payload Type is "negotiated". If I understand correctly,
> it's really NOT negotiated at all. *
>
> * *
>
> *I understand that static payload types, such as voice codecs (e.g.,PCMA,
> PCMU) are indeed negotiated. So my question then is with DTMF payload types.
> *
>
> * *
>
> *From my reading, it appears that a call request INVITE sender specifies a
> supported RFC 2833 DTMF RTP payload type by defining this RTP payload type
> with the specific 'telephone-event' rtpmap attribute in the SDP and by
> including a mapped payload type value (of the sender's choice) in the
> dynamic range (96-127) within the audio media stream 'm' line. As I
> understand, the SDP answerer can then respond to this request with its own
> custom mapped payload type value as long as it also defines the DTMF payload
> type with the specific 'telephone- event' rtpmap attribute in the SDP.*
>
> * *
>
> * *
>
> *So my question is what exactly does the RTP payload type number stand for
> within the confines of RFC 2833 & DTMF and the SIP UA which specifies the
> specific payload type number? Is this the payload type used by UA 1 (for
> example) to indicate which 8 bit number the SIP UA's peer (UA 2) is to use
> in the 7 bit RTP "Payload Type" header when sending DTMF (RFC 2833) to the
> UA 1? Again as I understand, each UA doesn't have to specify the same RTP
> payload type number for RFC 2833 DTMF to flow between the UAs. It appears
> two distinct payload types can be used in a single session - one traveling
> from UA 1 to UA2 and another from UA 2 to UA 1.*
>
> * *
>
> *Can someone confirm this understanding and correct any wrong assumptions?
> *
>
> * *
>
> *Thanks,*
>
> *Keshav Shrivastava*
>
> ** **
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20110830/2c87916f/attachment-0002.html>


More information about the discussion mailing list