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On 2/9/11 13:52, Feb 9, <a class="moz-txt-link-abbreviated" href="mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a> wrote:
<blockquote
cite="mid:DD8B10B86502AB488CB2D3DB4C546E38E6D160@008-AM1MPN1-004.mgdnok.nokia.com"
type="cite">
<div class="WordSection1"><span style="font-size: 11pt;
font-family: "Calibri","sans-serif";
color: rgb(31, 73, 125);">The RTP/media stack on the other
hand is definitely in the scope
of the IETF effort. I think we should standardize the RTP use
in the browsers
and that would be one step towards interop with SIP phones.
The critical thing
seems to be the STUN connectivity check or media authorization
part. If we
mandate browsers to get that exchange done before they are
allowed to generate
any RTP packets on behalf of the application, this will ruin
the possibility of
interop with 99% of existing SIP clients (without some kind of
an SBC). DTMF
transport capability may also be relevant interop requirement.<o:p></o:p></span><br>
</div>
</blockquote>
<br>
+1.<br>
<br>
Also, it would be incorrect to characterize RTP as being solely for
the benefit of SIP. Several other protocol -- both published and
proprietary -- make use of RTP; Jingle comes to mind, as do the
proprietary protocols used by various IP PBX vendors.<br>
<br>
If we decide to go down an "RTP but with special sauce" or "not
quite RTP" path, we lose a lot of potential functionality.<br>
<br>
/a<br>
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