[OpenSIPS-Users] RTP and RTCP

David Villasmil david.villasmil.work at gmail.com
Sat Aug 9 18:38:42 CEST 2008


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?

On Sat, Aug 9, 2008 at 2:52 PM, Ovidiu Sas <osas at voipembedded.com> wrote:

> You will need full support from the client and I doubt that there are
> implementation that are doing this.
>
> see http://www.ietf.org/rfc/rfc3605.txt:
>
> 2.1.  The RTCP Attribute
>
>   The RTCP attribute is used to document the RTCP port used for media
>   stream, when that port is not the next higher (odd) port number
>   following the RTP port described in the media line.  The RTCP
>   attribute is a "value" attribute, and follows the general syntax
>   specified page 18 of [RFC2327]: "a=<attribute>:<value>".  For the
>   RTCP attribute:
>
>   *  the name is the ascii string "rtcp" (lower case),
>
>   *  the value is the RTCP port number and optional address.
>
>   The formal description of the attribute is defined by the following
>   ABNF [RFC2234] syntax:
>
>   rtcp-attribute =  "a=rtcp:" port  [nettype space addrtype space
>                         connection-address] CRLF
>
>   In this description, the "port", "nettype", "addrtype" and
>   "connection-address" tokens are defined as specified in "Appendix A:
>   SDP Grammar" of [RFC2327].
>
>   Example encodings could be:
>
>    m=audio 49170 RTP/AVP 0
>    a=rtcp:53020
>
>    m=audio 49170 RTP/AVP 0
>    a=rtcp:53020 IN IP4 126.16.64.4
>
>    m=audio 49170 RTP/AVP 0
>    a=rtcp:53020 IN IP6 2001:2345:6789:ABCD:EF01:2345:6789:ABCD
>
>
> Regards,
> Ovidiu Sas
>
> On Sat, Aug 9, 2008 at 8:16 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > Yeah, that's exactly what I don't want. The idea is not to proxy media,
> let
> > media flow between the UACs, but proxy the RTCP...
> >
> >
> > thanks
> >
> > On Sat, Aug 9, 2008 at 2:12 PM, Adam Linford <adam.linford at oralnet.co.uk
> >
> > wrote:
> >>
> >> rtcp is sent to the exact same destination as the RTP, afaik, so if you
> >> proxy media in your calls, you could get ahold of those packets.
> >>
> >> Cheers,
> >> Adam
> >>
> >> On 9 Aug 2008, at 12:46, David Villasmil wrote:
> >>
> >>> Got an easy question:
> >>>
> >>>     RTCP packet are send to monitor media QoS, this much I know. ;) My
> >>> question is this: Are RTCP packets sent directly between end points? Or
> can
> >>> they be routed using a thrid party? For instance, Lets say 1 UAC makes
> a
> >>> call through SIP Server A, and UAC 2 ansers the call, RTP packets are
> sent
> >>> from UAC <--> UAC directly, but can they be instructed to send RTCP
> packets
> >>> via SIP Server A? I obviously haven't read the RFC, but if this could
> be
> >>> done, we would have a way of knowing whether the call is still up or
> not,
> >>> hence perfect accounting even if we don't receive the BYE from the
> UACs.
> >>>
> >>> thanks
> >>>
> >>>
> >>> david
> >>> _______________________________________________
> >>> Users mailing list
> >>> Users at lists.opensips.org
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20080809/9f99c16e/attachment-0001.htm 


More information about the Users mailing list