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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hello Abin Matthew/Brian Tate/Tomasz Czyzew/Dale Worley/Ellen Lee,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thank you everyone for the excellent feedback and response you
provided for our question. We have a good agreement within our team and we will
make the necessary changes. We see that the ACK of Re-INVITE has a valid SDP
and we should restart audio session according to the new SDP, this way we will
use the new Media IP and Port provided. Now the CSAT audio should be heard to
the caller.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Please mark this ticket as closed. Following are the response
that we received for the reference:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal>TICKET: Re-Invite without SDP (No Audio Issue)<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><b>RESPONSES FROM FORUM<o:p></o:p></b></p>
<p class=MsoNormal><b><o:p> </o:p></b></p>
<p class=MsoNormal><b>Abin Matthew:<o:p></o:p></b></p>
<p class=MsoNormal>So you are redirecting the caller to another application
which plays audio?(Yes) The reinvite should definitely have the new SDP.<o:p></o:p></p>
<p class=MsoNormal>It would actually depend on the call flow. Is the same gateway
playing the audio when application changes?(IVR plays the audio) If not then
SDP change is not required. Perhaps you need to look into the application, the
path defined for the media file and the gateway that plays it.<br>
When Agent enters extension and doesn't hear audio then it indicates an SDP
issue. There should be a new INVITE with SDP to setup the new RTP .i.e. Agent
hears enter agent I'd.<o:p></o:p></p>
<p class=MsoNormal>There should also be a reinvite back to the caller when
agent enter the extension to put the caller on hold. Usually a hold music is
played so that portion needs to be implemented on the application. When the
agent has entered the agent Id, there should be another reinvite with SDP to
connect the caller with the new destination.<br>
<br>
<b>Brian Tate:<o:p></o:p></b></p>
<p class=MsoNormal>1. no.<o:p></o:p></p>
<p class=MsoNormal>2. no.<o:p></o:p></p>
<p class=MsoNormal>“the re-INVITE will contain the offer” is “in regards
to audio.”<o:p></o:p></p>
<p class=MsoNormal>The SIP IVR should respond to the re-INVITE with a
non-faliure response (i.e. 1xx or 2xx code), and contain an SDP offer. <o:p></o:p></p>
<p class=MsoNormal>And the SIP IVR should expect the subsequent ACK sent back
from the Avaya to contain the SDP answer, based on the offer/answer model
detailed in RFC 3264.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><b>Tomasz Czyzew:<o:p></o:p></b></p>
<p class=MsoNormal>I am not sure if I understand correctly the call scenario
you described, but I think that the UAS should respond with provisional 1xx
containing SDP for this session, so none of the above answers is correct.<o:p></o:p></p>
<p class=MsoNormal>SDP-less INVITE is sent by UAC to "pull the SDP out of
the UAS". In practice SDP-less INVITE is used for third party call control
(3PCC) - to establish a call in someone else's behalf.<o:p></o:p></p>
<p class=MsoNormal>Please take a look into RFC6337 (3.1.2. INVITE Request
without SDP).<o:p></o:p></p>
<p class=MsoNormal><b>Dale R Worley:<o:p></o:p></b></p>
<p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>The
SIP IVR must (in its 200 response to the re-INVITE) send an SDP offer. This
SDP must be a legitimate update of the SDP that it previously sent within the
dialog; in particular, media format numbers that have been used previously must
not be re-used for different codecs than before. It should also offer all
codecs and media types that it is willing to support (at this time), not just
the ones that it is using in the current media session.<o:p></o:p></span></p>
<p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>The
peer will send an ACK, which must contain an SDP answer to the SDP offer that
the SIP IVR sent in the 200. At this point, the SIP IVR must reconfigure its
media processing to utilize the newly-negotiated media session.<o:p></o:p></span></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p> </o:p></span></b></p>
<p class=MsoNormal><b>Ellen Lee:<o:p></o:p></b></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:black'>Re-Invite is to put caller on/off hold during call transfer. Not
hearing the CSAT survey indicates SIP/RTP setup issues between Avaya and IVR.
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p></o:p></span></p>
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