e offer SDP was
> received in INVITE / ReINVITE 200 response.
>
> 3PCC services are a good use case of the above.
>
> Regards
> Tarun Gupta
> Aricent
>
> -Original Message-
> From: ikuzar RABE [mailto:ikuzar9...@gmail.com]
> Sent: Wednesday, July 24, 2013 7:46
Hi all,
I would like to know in which cases a caller include its media capabilities
(SDP) in ACK instead in INVITE method ...
Thanks for your help,
ikuzar.
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Ok, thanks for these responses. I 'd like to add this:
In RFC 3261 page 38 it is said:
Use of cryptographically random identifiers (RFC 1750 [12]) in the
generation of Call-IDs is RECOMMENDED.
in RFC 1750 8.2.3 it is said :
probably a good minimum for a very
high security cryptographic ke
Hi all,
I found with the link below (IBM) that Call-id max length is 256
characters:
http://pic.dhe.ibm.com/infocenter/wvraix/v6r1m0/index.jsp?topic=%2Fcom.ibm.wvraix.voip.doc%2Fsiptags.html
Microsoft said the same:
http://msdn.microsoft.com/en-us/library/ff595864%28v=office.12%29.aspx.
1) Why do
it has violated
> the rules for a proxy. Devices that do this are typically called Session
> Border Controllers. It is very common. There are both advantages and
> disadvantages to doing this.
>
> Thanks,
> Paul
>
> On 7/17/13 5:29 AM, ikuzar RABE wrote:
>
Hi all,
I saw a RTP flow which is not directly established between UAC and UAS but
goes through a SIP proxy ...
Is there any information in SIP message exchange producing this situation ?
Thanks for your help,
ikuzar
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Hi all,
I'd like to know if there are SIP messages exchange to maintain the session
between two UA (message such as "Hello, I'am still here" / "Hello, I'am
still here too" while RTP conversation is established and lasts a longtime.
The context:
I have to differentiate an INVITE with a BYE which a
connect party A to party C.**
> **
>
> ** **
>
> ** **
>
> *From:* ikuzar RABE [mailto:ikuzar9...@gmail.com]
> *Sent:* Tuesday, June 18, 2013 11:36 AM
> *To:* Johan DE CLERCQ
> *Cc:* Brett Tate; sip-implementors@lists.cs.columbia.edu
>
> *Subject:* Re: [Sip-implementors] ports renegociatio
gt; Sent: dinsdag 18 juni 2013 14:14
> To: ikuzar RABE; sip-implementors@lists.cs.columbia.edu
> Subject: Re: [Sip-implementors] ports renegociation - Email found in
> subject
>
> Yes; it is possible. It can occur anytime a device wants to do it. Some
> B2BUAs will do it (change bo
Nobody has an idea ... ??
2013/6/11 ikuzar RABE
> Hi all,
>
> I 'd like to know if ports can be renegotiated (with a new INVITE for
> example) while RTP flows are already established ... !
> If it is possible, in which case we can face with this situation ?
> (multicas
Hi all,
I 'd like to know if ports can be renegotiated (with a new INVITE for
example) while RTP flows are already established ... !
If it is possible, in which case we can face with this situation ?
(multicast ? other ?)
Thanks for your help
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Hi all,
I work on Ubuntu 10.4.
I parse the SIP message with strtok_r (My SIP message parser must be
reentrant). The function prototype is like this:
char *strtok_r(char *str, const char *delim, char **saveptr);
I put this function in a while loop. Each iteration, *saveptr* points to
the next lin
Hello,
*
*
I'd like to know how can I retrieve IP address for RTP flow from SIP
message ?
I can obtain IP address in Contact field, but it may be also a domain name
instead of IP address...
Thanks for your help.
ikuzar
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Hi all,
I 'd like to retrieve IP address of INVITE sender. Today I retrieve it from
Contact header field. But is it a good idea ? What about retrieving it from
Message Body (from creator/owner (o) or connection information (c) ) ?
thanks for your help
_
Hi all,
I'd like to know what are the differences between Request-URI and to URI ?
from URI and contact URI ?
Thanks for your help
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