[SIPForum-discussion] SIP Flow with a 200OK(without SDP) to the INVITE

Fri Jun 21 08:30:48 UTC 2019

Hi Guillermo,

This flow looks just fine and is very common especially in mobile networks,
where a bearer is being established before the B party answers. This allows
to play an early media to the A-party in first place, but together with
UPDATE/200, it also allows both parties to renegotiate the codec even
before the call is answered (e.g. in CSFB scenarios).

Take a look at RFC3261:
"

13.2.1 <https://tools.ietf.org/html/rfc3261#section-13.2.1> Creating
the Initial INVITE

[...]

      o  If the initial offer is in an INVITE, the answer MUST be in a
         reliable non-failure message from UAS back to UAC which is
         correlated to that INVITE.  For this specification, that is
         only the final 2xx response to that INVITE.  That same exact
         answer MAY also be placed in any provisional responses sent
         prior to the answer.  The UAC MUST treat the first session
         description it receives as the answer, and MUST ignore any
         session descriptions in subsequent responses to the initial
         INVITE.

"
For your case the "reliable non-failure message from UAS back to UAC which
is correlated to that INVITE" is this 183 with SDP, because is is followed
by PRACK+200. There should be no more SDP answers after this provisional
(and reliably delivered) reply. You can find more on this topic in RFC3262:
"

5 <https://tools.ietf.org/html/rfc3262#section-5> The Offer/Answer
Model and PRACK

   If the INVITE contained an offer, the UAS MAY generate an answer in a
   reliable provisional response (assuming these are supported by the
   UAC).  That results in the establishment of the session before
   completion of the call.
"

This call flow was also described in RFC 6337 as "Pattern 3".

P.S.: To be able to try to help you with the audio issue, I would need to
look into pcap trace of the call.

BR,
Tomasz.

czw., 20 cze 2019 o 19:41 Zuñiga, Guillermo <Guillermo.Zuniga at cwpanama.com>
napisał(a):

>
>
> Hi fellows,
>
>
>
> I would like to know the following:
>
>
>
> I am having the following flow and I would like to know if is a valid flow
> and if the remote SIPGW should be able to handle or complete this dialog
> when receive a 200OK without SDP to the INVITE.
>
>
>
> Flow:
>
> A ->INVITE with SDP
>
> B<-TRYING
>
> B<-183 WITH SDP
>
> A->PRACK
>
> B<-200OK TO THE PRACK
>
> B<-180 RINGING WITHOUT SDP
>
> B<-200OK WITHOUT SDP TO THE INVITE.
>
> A->ACK TO THE 200OK WITHOUT SDP TO THE INVITE.
>
>
>
>
>
> My question comes, cause I am having audio issue with the client is
> calling us and here I would like to know if could be related that I am
> sending the 200OK without SDP. Or he should be able to complete the dialog
> cause previous to the 200OK without SDP I sent a 183 with SDP?
>
>
>
> Please let me know any comments or where in the RFC I  can look for that
> describe this behavior.
>
>
>
> Regards,
>
> Guillermo
>
>
>
>
>
> Guillermo Zuniga
> Especialista de Soporte Técnico
> Gerencia de Soporte Técnico
>
> P: +507 263-6671 | +507 6670-0481
> E: Guillermo.Zuniga at cwpanama.com |W: www.cwpanama.com
>
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