[SIPForum-discussion] Incorrect RTCP reports being generated.

[usman chaudhry]

Hello All,

    I have a peculiar issue where for a SIP --> PSTN call the RTCP reports (i.e. Sender Reports) generated by SIP UA (i.e. xlite) count all RTP packets that come before the 200 OK as lost. These RTCP reports are then used to calculate the MoS for the call and since the SIP UA reports these packets as lost the MoS value is abysmally low (i.e. 1.0 - 2.0) whereas in reality the call quality is all right . Note that these RTP packets (highlighted in RED) are sent by the MGW to playback the Ring tone generated by the PSTN; and as far as I know this is the correct call flow for a SIP-->PSTN Call. Following is the basic call flow.


SIP (UA)                                    MGW                                   PSTN
 
----------------------INVITE---------------------->
<---------------- 100 
Trying --------------------                                                       
                                                         ------------------ IAM 
--------------------->
                                                         <---------------- 
ACM --------------------
                                                         
<--------- PSTN Ringing Tone ------
<------------Ring Tone (RTP) 
-----------------
<-------- 183 Session In Progress 
--------
                                                        
<---------------- ANM --------------------
<---------------- 200 
OK ------------------------
 
<------------------------------------ Voice Path 
complete ---------------------------------->  
    Now I have the following question with this regard
1) Is this the correct behavior by the SIP UA (i.e. xlite) to report these  RTP packets as dropped. Note that the SIP UA uses these packets to playback the ringtone.
2) Is there any standard which specifies  how to generate RTCP reports for this particular call flow. Note that network uses RFC 1889 for RTP/RTCP traffic

Rgds
Usman.


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20081008/d426c3a1/attachment-0002.html>