[SIPForum-discussion] one way communication

[SINGH ANURAG]

hello All,

ethereal trace cannot be attached due to size limits

the call there is gap in the rtp flow from UA2 to UA1.
after that all packet are freed by the jitter for gateway.

can you please let me know 
--is this gap acceptable and rtp flow should resume after that. 
--jitter is correct in saying that to all packets "packet comes too
late" after this gap.
-- or jitter should Only say to that to the first packet and accept the
later ones. 

regards
Anurag

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of SINGH ANURAG
Sent: Friday, January 23, 2009 5:27 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] one way communication

Hello All,

UA1
UA2
--INVITE with SDP (g711 ptime 20 mxptime=90)--------------->
 <----183 progress with SDP (g711 ptime=20 mxptime=30---------
 <----200OK with SDP (g711 ptime=20 mxptime=30---------
<---------------------RTP------------------------------
----------------------RTP------------------------------->

This is wireshark information however UA1 cannot hear anything after
200OK.

for information : I can see UA1 jitter modules throw a error "packet
comes too late" for all rtp packets. however wireshark shows no packet
delay or loss packet

can you please provide input what could be the reason?
Thanks in advance for help

Regards
Anurag

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