[OpenSIPS-Users] RTP and RTCP

[Ovidiu Sas]

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
>>
>
>
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