[SIPForum-discussion] 5 minute cut-off

Jorge jwalcantara at hotmail.com
Tue Sep 3 12:14:18 UTC 2013


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 <mailto:smraney at comcast.net> "
<smraney at comcast.net <mailto:smraney at comcast.net> >
To: Tristan Bulandus <tristanbulandus at yahoo.com
<mailto:tristanbulandus at yahoo.com> > 
Cc: Jorge <jwalcantara at hotmail.com <mailto:jwalcantara at hotmail.com> >; Binan
AL Halabi <binanalhalabi at yahoo.com <mailto:binanalhalabi at yahoo.com> >;
murali87ece at gmail.com <mailto:murali87ece at gmail.com> ;
discussion at sipforum.org <mailto: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
<mailto:tristanbulandus at yahoo.com> >
To: "Jorge" <jwalcantara at hotmail.com <mailto:jwalcantara at hotmail.com> >,
"Binan AL Halabi" <binanalhalabi at yahoo.com <mailto:binanalhalabi at yahoo.com>
>, smraney at comcast.net <mailto:smraney at comcast.net> , murali87ece at gmail.com
<mailto:murali87ece at gmail.com> 
Cc: discussion at sipforum.org <mailto: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! 

 

 

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