thanks you all.
On Wed, May 22, 2013 at 2:56 PM, Olle E. Johansson wrote:
>
> 22 maj 2013 kl. 19:45 skrev Brett Tate :
>
> >>> Hello. I did that, but I dont have some docs
> >>> to compare and determine is what the client
> >>> is doing is correct or not,
> >>
> >> http://tools.ietf.org/html/rf
22 maj 2013 kl. 19:45 skrev Brett Tate :
>>> Hello. I did that, but I dont have some docs
>>> to compare and determine is what the client
>>> is doing is correct or not,
>>
>> http://tools.ietf.org/html/rfc3261
>>
>> The ultimate reference to SIP.
>> /O
>
> Since RFC 3261 doesn't provide muc
> > Hello. I did that, but I dont have some docs
> > to compare and determine is what the client
> > is doing is correct or not,
>
> http://tools.ietf.org/html/rfc3261
>
> The ultimate reference to SIP.
> /O
Since RFC 3261 doesn't provide much guidance concerning when to
open/reuse/close a TCP
thanks much
On Wed, May 22, 2013 at 2:26 PM, Olle E. Johansson wrote:
>
> 22 maj 2013 kl. 19:20 skrev Milton Sanchez :
>
> > Hello. I did that, but I dont have some docs to compare and determine is
> > what the client is doing is correct or not,
> http://tools.ietf.org/html/rfc3261
>
> The ulti
22 maj 2013 kl. 19:20 skrev Milton Sanchez :
> Hello. I did that, but I dont have some docs to compare and determine is
> what the client is doing is correct or not,
http://tools.ietf.org/html/rfc3261
The ultimate reference to SIP.
/O
> thanks.
>
>
> On Wed, May 22, 2013 at 1:55 PM, Nahum Nir
Hello. I did that, but I dont have some docs to compare and determine is
what the client is doing is correct or not,
thanks.
On Wed, May 22, 2013 at 1:55 PM, Nahum Nir wrote:
> I would wireshark some client.
>
>
> On Mon, May 20, 2013 at 10:49 PM, Milton Sanchez
> wrote:
>
>> Hello, can someone
I would wireshark some client.
On Mon, May 20, 2013 at 10:49 PM, Milton Sanchez wrote:
> Hello, can someone please point me to the right direccion on finding some
> infomarction/tutorial about SIP over TCP.
>
> regards
> Milton
> ___
> Sip-implementors
Hello, can someone please point me to the right direccion on finding some
infomarction/tutorial about SIP over TCP.
regards
Milton
___
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2011/1/19 Andreas Byström (Polystar T & M) :
> For some reason UAB don't accept the incoming BYE. I see that during these 10
> seconds, UAA takes down the TCP connection so it creates a new one before
> sending BYE. Is this allowed? Should UAB side be able to handle this?
>
> If I instead let the
...@wipro.com [mailto:santosh.kalsang...@wipro.com]
Sent: den 19 januari 2011 15:16
To: Andreas Byström (Polystar T & M); sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] SIP over TCP
Hi Andreas,
Is it 481 (call leg/transaction does not exist) response for BYE? I happen to
Behalf Of Andreas
Byström (Polystar T & M)
Sent: Wednesday, January 19, 2011 6:23 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] SIP over TCP
Hi,
I've just recently started to look at a scenario when SIP is sent over TCP (not
UDP as I'm used to have). I
Hi,
I've just recently started to look at a scenario when SIP is sent over TCP (not
UDP as I'm used to have). I see a behavior that I'm not really sure it correct:
It's a very basic SIP call UAA -> UAB
1. UAA sends INVITE to UAB
2. UAB answers with 180 and 200 OK
3. UAA send
On Wed, 2010-01-20 at 14:22 +0800, LIU Liping wrote:
> When TCP is used as transport lay for SIP messages. Maybe the SIP
> Stack can read some data from socket which includes more than one sip
> message and maybe the SIP stack can read only a part of a sip message.
> So, now Is it the stack's duty
El Miércoles, 20 de Enero de 2010, LIU Liping escribió:
> hi all,
>When TCP is used as transport lay for SIP messages. Maybe the SIP
> Stack can read some data from socket which includes more than one sip
> message and maybe the SIP stack can read only a part of a sip message.
> So, now Is it t
hi all,
When TCP is used as transport lay for SIP messages. Maybe the SIP
Stack can read some data from socket which includes more than one sip
message and maybe the SIP stack can read only a part of a sip message.
So, now Is it the stack's duty to determine the sip message boundaries
by the ST
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