[SIPForum-discussion] Call Hold

Donald Lee baolovebao at gmail.com
Thu Jan 10 08:29:05 UTC 2008


Please refer to to comments inline.

On Jan 8, 2008 8:44 PM, Halit Sakca <sakcahalit at hotmail.com> wrote:
>
> Hi,
Hi All,

Suppose there is SIP session established between 2 SIP UA ('A' & 'B).

Now 'A' put 'B' on Hold by sending REINVITE with sendonly.

1) Please let me know , it is follow the latest standard of RFC
# I think it has not been standarded yet. But it is a normal procedure to
Call hold.
See
http://ietf.org/internet-drafts/draft-ietf-sipping-service-examples-13.txtfor
more information. There is another method is using connection
parameter
in SDP as the indicator, but it's not recommended yet now.

2) In case only direction RTP stops or both ?
# B's direction is now recvonly, it doesn't send any RTP to A. For A,
sendonly.

3) In this if the internediate server(proxy server) is down during put on
hold condition, user will re-transmit re-INVITE till 31 sec. In this case
wht is the status of the call from both end.
# re-INVITE failure doesn't teardown the dialog. remaining the hold status.

4) When user 'A' put on hold by sending INVITE , does he stop sending the
RTP packet parallely to the sending re-INVITE
# It depends on the implementation. RTP stops after receiving the final
response of re-INVITE in theroy.

Thanks in Advance
>
>  Also I would like to add;
>
>  -- gatekeeper deals with call session establishment, gateway is not able
to
> establish a sip call.
>  one exapmle;
>  A calls B,
>  A:pots client,
>  B:sip client,
>
>  [A] --(megaco)-->[Gateway]--(SIP)--->[Gatekeeper]---(SIP)---> [B]
>
> hopefully more clear,
>
> Selamlar,
> Halit Sakca
>
>
>
>
>  ________________________________
>  From: pinakee.b at xius.com
> To: asitdash1982 at gmail.com; discussion at sipforum.org
> Date: Tue, 8 Jan 2008 16:04:46 +0530
>
> Subject: Re: [SIPForum-discussion] Call Hold
>
>
>
>
> Asit,
>
>
>
> These are H.323 components and not part of SIP. In H.323, Gatekeeper deals
> with registration and location. Gateway deals with Call Control and media.

>
>
>
>  ________________________________
>
>
> From: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] On Behalf Of asit dash
> Sent: Tuesday, January 08, 2008 12:52 PM
> To: discussion at sipforum.org
> Subject: Re: [SIPForum-discussion] Call Hold
>
>
>
> Hi all,
>     can any one tell me the difference between gateway and gatekeeper  ?
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-- 
BR
Donald
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