Hi Richard,

thanks for your answer.

I totally agree to you when you interprete this as a change of payload type
number from 118 to 8.
But I think it is also a legitimate view, that this Re-Invite does not
represent a change of payload type number, but a change of the attributes
of payload 8.

So in this case, parameters of codec 118 did not change, but only the
parameter of 8 changed.
Is this actually possible that different payload type numbers have the same
attributes?


Best Regards
Philipp Schöning


Am Mo., 29. Okt. 2018 um 08:06 Uhr schrieb Richard Phernambucq <
r.phernamb...@cuperus.nl>:

> Hi Philipp,
>
> According to RFC 3264 the server sending the Re-Invite is wrong. See
> chapter 8.3.2 (Changing of Set of Media Formats):
>
> (...) in the case of RTP, the mapping from a particular dynamic payload
> type
>     number to a particular codec within that media stream MUST NOT change
>     for the duration of a session.
>
> So changing the payload type number from 8 to 118 during the session is
> not allowed. It would have been allowed though if payload type number
> 118 wasn't already in use (see same chapter):
>
> (...) it is acceptable for multiple payload type numbers to be mapped to
> the same
>     codec, so that an updated offer could also use payload type number 72
>
>
>
> Best regards,
> Richard Phernambucq
>
>
>
> On 27-10-2018 14:52, Philipp Schöning wrote:
> > Hi all,
> >
> > we saw a SIP implementation violating the SDP offer/answer procedure when
> > answering a Re-Invite.
> >
> > Here is the signalling:
> >
> > Server                                       Client
> >
> >
> >
> >     +----------------INVITE--------------------->
> >
> >     m=audio 18410 RTP/AVP 8 100 118
> >
> >     b=AS:87
> >
> >     b=RS:1087
> >
> >     b=RR:3262
> >
> >     a=rtpmap:8 PCMA/8000
> >
> >     a=fmtp:8 vad=no
> >
> >     a=rtpmap:100 telephone-event/8000
> >
> >     a=fmtp:100 0-15
> >
> >     a=rtpmap:118 PCMA/8000
> >
> >     a=gpmd:118 vbd=yes
> >
> >     a=sendrecv
> >
> >
> >
> >     <----------------200 OK---------------------+
> >
> >                 m=audio 9158 RTP/AVP 8 118 100
> >
> >                 a=rtpmap:8 PCMA/8000
> >
> >                 a=fmtp:8 vad=no
> >
> >                 a=rtpmap:118 PCMA/8000
> >
> >                 a=rtpmap:100 telephone-event/8000
> >
> >                 a=fmtp:100 0-15
> >
> >                 a=sendrecv
> >
> >
> >
> >     +-----------------Re-Invite----------------->
> >
> >     m=audio 18410 RTP/AVP 8
> >
> >     a=rtpmap:8 PCMA/8000
> >
> >     a=gpmd:8 vbd=yes
> >
> >     a=silenceSupp:off - - - -
> >
> >
> >
> >     <----------------200 OK---------------------+
> >
> >                          m=audio 9158 RTP/AVP 118
> >
> >                          a=rtpmap:118 PCMA/8000
> >
> >                          a=sendrecv
> >
> >
> > In the last 200OK Client is sending a Payload type (118) which was not
> > offered in the Re-Invite. The Re-Invite is generated when the SIP Server
> is
> > detecting a fax transmission and applies ITU V.152 (Voice Band Data),
> this
> > is also how gpmd Attribute is introduced.
> >
> >
> > The SIP Client vendor commented this on the case:
> >
> > *"after call is answered (media session is established), provider
> > renegotiates codec set and changes PCMA codec description bound to
> payload
> > 8 with the description which was bound to payload 118 in initial
> > negotiation of media session.*
> >
> > *It causes conflict, because new description of PCMA codec is already
> bound
> > to payload 118 in initial offer/answer.*
> >
> >
> > *Voip provider can configure own side of trunk and avoid usage of
> > non-standard extensions in payload description (remove unnecessary
> > duplication of codecs). Also, provider need to preserve payload
> definitions
> > when renegotiate media session."*
> >
> >
> > So is the vendor right by saying that payload description have to be
> > preserved during a session? It did not find this in the SDP offer/answer
> > RFCs.
> >
> > We never saw any other vendor having issues with this signalling.
> >
> >
> > Best Regards
> > Philipp Schöning
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@lists.cs.columbia.edu
> > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
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