[SIPForum-discussion] 5 minute cut-off

[bikram mohanty]

I think there is problem while negotiating Re-Invite sent from any device for Session-Refresh.
That ,some where you know could be getting some error and that's why the call is dropped  after 5 minutes.

You can identify Session refresh INVITE either in the tag in To filed.
or Session expire related parameter in that INVITE.

if you find out from where the Session-Refresh INVITE is coming you can also modify the Session expire to bigger value to understand this problem is caused by the Refresh  and the Re-Invite not being honored properly.

Thanks





________________________________
 From: Jorge <jwalcantara at hotmail.com>
To: 'Tristan Bulandus' <tristanbulandus at yahoo.com>; smraney at comcast.net 
Cc: discussion at sipforum.org 
Sent: Tuesday, 3 September 2013 5:44 PM
Subject: Re: [SIPForum-discussion] 5 minute cut-off
 


ptime has to do with media/RTP packetization time , in other words a ptime of 20 says that the media will be sent in 20 millisecond carrying  payload/voice packets of RTP or 50 packets per second.  This is not the cause of the you hang up problem.
 
The problem has to do with re-INVITES and how often they are sent by the varying end points.
An INVITE with a tag in the TO: field is a re-INVITE.
 
 
Jorge Alcantara
 
From:Tristan Bulandus [mailto:tristanbulandus at yahoo.com] 
Sent: Tuesday, September 03, 2013 1:50 AM
To: smraney at comcast.net
Cc: Jorge; Binan AL Halabi; murali87ece at gmail.com; discussion at sipforum.org
Subject: Re: 5 minute cut-off
 
Hi.
 
Huawei softswitch sends BYE with reason: SESSION TIMER EXPIRES. Our Huawei Softswitch has Timer of 300s which is why calls for IOS and Android gets cut off at around 5 minutes.
 
My only concern is that 3 of the other SIP Clients do not experience a call cut-off at around 5 minutes while simulating the exact customer scenario.
 
Would you be familiar on the function of this 'a=ptime' attributes? Is the ptime value in seconds? Would appreciate an immediate reply please.
 
Thanks.
 
From:"smraney at comcast.net" <smraney at comcast.net>
To: Tristan Bulandus <tristanbulandus at yahoo.com> 
Cc: Jorge <jwalcantara at hotmail.com>; Binan AL Halabi <binanalhalabi at yahoo.com>; murali87ece at gmail.com; discussion at sipforum.org 
Sent: Monday, September 2, 2013 8:50 PM
Subject: Re: 5 minute cut-off
 
Tristan,
 
I admittedly do have have working experience with all these particular endpoints, so my suggestions are merely based on observations of the Invite messages.
 
The Invite's associated to the IOS and Android endpoints do not appear to have 'ptime' attributes, whereas the Invites associated to the other three endpoints do.  Issues with ptime attributes typically effect quality of service.
 
PTimer Attributes:
 a. Genband IP Phone - a=ptime: 20
 b. Grandstream IP Phone - a=ptime: 20
 c. IOS - Not Configured
 d. Android - Not Configured
 e. PC Client - a=ptime: 30
 
My other thought was whether or not this could be associated to a 'session refresh' issue, if the session is seen as inactive and requires refresh to remain established?
 
Thanks,
 
Steve.....
 
From: "Tristan Bulandus" <tristanbulandus at yahoo.com>
To: "Jorge" <jwalcantara at hotmail.com>, "Binan AL Halabi" <binanalhalabi at yahoo.com>, smraney at comcast.net, murali87ece at gmail.com
Cc: discussion at sipforum.org
Sent: Sunday, September 1, 2013 11:33:43 PM
Subject: 5 minute cut-off
 
Hi guys.
 
Genband A2 is currently peered to Huawei Softswitch.
 
Genband A2 has the following terminals that register to it:
 a. Genband IP Phone
 b. Grandstream IP Phone
 c. IOS
 d. Android
 e. PC Client
 
Now, I try to call to Numbers routed at the Huawei Softswitch from the following terminals and experience:
a. Genband IP Phone calling Huawei - Call NOT cut-off
b. Grandstream IP Phone calling Huawei - Call NOT cut-off
c. IOS calling Huawei - Call cut-off at 5 minutes
d. Android calling Huawei - Call cut-off at 5 minutes
e. PC Client calling Huawei - Call NOT cut-off
 
May I request what has to be adjusted on the IOS and Android clients please? I've attached a summary of the SIP Invites being generated by terminals.
 
Thanks! 
 
 
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20130905/b834c38c/attachment-0002.html>