[SIPForum-discussion] Incorrect RTCP reports being generated.

[usman chaudhry]

Hello All,

    I have noticed that the RTP stream that is coming from the MGW to the CPE has its sequence number being reset twice. Can this be the reason for the incorrect RTCP Senders Reports generated by the CPE. Refer to the attached JPEG, where 192.168.1.4 is the MGW and 10.11.128.126 is the CPE. Note that the Seq Number changes but the SSRC remains the same. Is this allowed as per RFC 1889; its not very clear.
   

Regards
Usman.




----- Original Message ----
From: "Beith, Gordon" <GBeith at empirix.com>
To: usman chaudhry <usman_chaudhry at yahoo.com>
Sent: Wednesday, October 8, 2008 10:47:40 PM
Subject: RE: [SIPForum-discussion] Incorrect RTCP reports being generated.

 
Sounds wrong to me. Seems that the Sip UA
has an odd implementation.
/Gordon
 

________________________________
 
From:discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of usman chaudhry
Sent: Wednesday, October 08, 2008
3:01 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion]
Incorrect RTCP reports being generated.
 
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.


      
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