<div dir="ltr">So if I understand this completely, when the proxy sends the SDP, if our proxy (openSIPS) sends our mediaproxy/rtpproxy IP/PORT but the endpoint's IP/PORT, media goes directly to the UAC but RTCP goes via out mediaproxy/rtpproxy... if the UAC if rfc3605 compliant?<br>
<br><div class="gmail_quote">On Sat, Aug 9, 2008 at 2:52 PM, Ovidiu Sas <span dir="ltr"><<a href="mailto:osas@voipembedded.com">osas@voipembedded.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
You will need full support from the client and I doubt that there are<br>
implementation that are doing this.<br>
<br>
see <a href="http://www.ietf.org/rfc/rfc3605.txt" target="_blank">http://www.ietf.org/rfc/rfc3605.txt</a>:<br>
<br>
2.1. The RTCP Attribute<br>
<br>
The RTCP attribute is used to document the RTCP port used for media<br>
stream, when that port is not the next higher (odd) port number<br>
following the RTP port described in the media line. The RTCP<br>
attribute is a "value" attribute, and follows the general syntax<br>
specified page 18 of [RFC2327]: "a=<attribute>:<value>". For the<br>
RTCP attribute:<br>
<br>
* the name is the ascii string "rtcp" (lower case),<br>
<br>
* the value is the RTCP port number and optional address.<br>
<br>
The formal description of the attribute is defined by the following<br>
ABNF [RFC2234] syntax:<br>
<br>
rtcp-attribute = "a=rtcp:" port [nettype space addrtype space<br>
connection-address] CRLF<br>
<br>
In this description, the "port", "nettype", "addrtype" and<br>
"connection-address" tokens are defined as specified in "Appendix A:<br>
SDP Grammar" of [RFC2327].<br>
<br>
Example encodings could be:<br>
<br>
m=audio 49170 RTP/AVP 0<br>
a=rtcp:53020<br>
<br>
m=audio 49170 RTP/AVP 0<br>
a=rtcp:53020 IN IP4 <a href="http://126.16.64.4" target="_blank">126.16.64.4</a><br>
<br>
m=audio 49170 RTP/AVP 0<br>
a=rtcp:53020 IN IP6 2001:2345:6789:ABCD:EF01:2345:6789:ABCD<br>
<br>
<br>
Regards,<br>
<font color="#888888">Ovidiu Sas<br>
</font><div><div></div><div class="Wj3C7c"><br>
On Sat, Aug 9, 2008 at 8:16 AM, David Villasmil<br>
<<a href="mailto:david.villasmil.work@gmail.com">david.villasmil.work@gmail.com</a>> wrote:<br>
> Yeah, that's exactly what I don't want. The idea is not to proxy media, let<br>
> media flow between the UACs, but proxy the RTCP...<br>
><br>
><br>
> thanks<br>
><br>
> On Sat, Aug 9, 2008 at 2:12 PM, Adam Linford <<a href="mailto:adam.linford@oralnet.co.uk">adam.linford@oralnet.co.uk</a>><br>
> wrote:<br>
>><br>
>> rtcp is sent to the exact same destination as the RTP, afaik, so if you<br>
>> proxy media in your calls, you could get ahold of those packets.<br>
>><br>
>> Cheers,<br>
>> Adam<br>
>><br>
>> On 9 Aug 2008, at 12:46, David Villasmil wrote:<br>
>><br>
>>> Got an easy question:<br>
>>><br>
>>> RTCP packet are send to monitor media QoS, this much I know. ;) My<br>
>>> question is this: Are RTCP packets sent directly between end points? Or can<br>
>>> they be routed using a thrid party? For instance, Lets say 1 UAC makes a<br>
>>> call through SIP Server A, and UAC 2 ansers the call, RTP packets are sent<br>
>>> from UAC <--> UAC directly, but can they be instructed to send RTCP packets<br>
>>> via SIP Server A? I obviously haven't read the RFC, but if this could be<br>
>>> done, we would have a way of knowing whether the call is still up or not,<br>
>>> hence perfect accounting even if we don't receive the BYE from the UACs.<br>
>>><br>
>>> thanks<br>
>>><br>
>>><br>
>>> david<br>
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>><br>
><br>
><br>
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><br>
</div></div></blockquote></div><br></div>