[Asterisk-Dev] [Fwd: Re: [Sip-implementors] Call hold questions]

[Olle E. Johansson]

For your information... I don't know how call hold is implemented in sip channels today.

/O

-------- Original Message --------
Subject: Re: [Sip-implementors] Call hold questions
Date: Tue, 25 Nov 2003 11:22:51 -0500
From: Mark A. Stacy <mstacy at delta-info.com>
Organization: Delta Informations Systems, Inc.
To: Andreas Byström <andreas.bystrom at e-horizon.se>,   "Sip Implemators" <sip-implementors at cs.columbia.edu>
References: <OIEIIHLKNIBLFMPEIPLJOEMJCKAA.andreas.bystrom at e-horizon.se>

Hello Andreas,

At SipIT 13 in Ottawa method #2 was discussed as the preferred mechanism.

Regards,
Mark

----- Original Message -----
From: "Andreas Byström" <andreas.bystrom at e-horizon.se>
To: "Sip Implemators" <sip-implementors at cs.columbia.edu>
Sent: Tuesday, November 25, 2003 11:20 AM
Subject: [Sip-implementors] Call hold questions


> I'm a bit confused regarding Call Hold service using SIP. I have seen two
> differnet approaches to solve this:
>
> 1. Send a re-Invite with "0.0.0.0" as the IP address in the sdp data
> 2. Send a re-Invite with the parameter a=sendonly set in the sdp data
>
> Is there some draft or RFC about this? Which one is preferred? I guess it
is
> best to have support for both ways (at least receiveng call hold) but when
I
> send the call hold I need to know which way to use.
>
> Do you see any pros/cons about the two different solutions?
>
> Regards,
> Andreas
>
> ____________________________________
> Andreas Bystrom
> Computer Engineer
>
> e-horizon Streaming Technologies
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>
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-- 
*** Olle E. Johansson, oej at edvina.net

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