[OpenSIPS-Users] RTP and RTCP

[Ovidiu Sas]

Yes, if both clients have support for rfc3506 you will be able to
proxy the RTCP stream only and leave the RTP stream fly between
endpoints.
Or, in principle, if only one part has support for rfc3506, you should
be able to proxy ony one RTCP stream (not both).

Regards,
Ovidiu Sas

On Sat, Aug 9, 2008 at 12:52 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> So something like this SHOULD do the trick?
>
> v=0
> o=alice 2890844526 2890844526 IN IP4 user.address.com  <------ USERS IP FOR
> RTP
> s=
> c=IN IP4 host.atlanta.example.com
> t=0 0
> m=audio 49170 RTP/AVP 0 8 97
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> m=video 51372 RTP/AVP 31 32
> a=rtpmap:31 H261/90000
> a=rtpmap:32 MPV/90000
> m=audio 49170 RTP/AVP 0
> a=rtcp:53020 IN IP4 rtp-proxy.address.com          <---- OUR RTP PROXT
>
> If the UAC is rfc compliant, then rtps would flow directly from UAC to UAC,
> *BUT* RTCP would go via our rtpproxy/mediaproxy.
>
> If this does work this way, we can -at least in principle- do accurate
> accounting without having to be in the middle or rtps flow! Can you imagine
> the cost-savings this would entitle???
>
> we need an rtp expert here...
>
> ;-)
>
> David
>
> On Sat, Aug 9, 2008 at 6:38 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> 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
>>> >
>>> >
>>
>
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