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List: sip-implementors
Subject: Re:RE: [Sip-implementors] 183 response
From: Lorenzo_Boffelli () alliedtelesyn ! com
Date: 2002-02-21 17:47:29
Message-ID: 00294BEE.C21145 () alliedtelesyn ! com
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Subject: RE: [Sip-implementors] 183 response
Author: Attila Sipos <AttilaS@vegastream.com>
Date: 2/20/2002 2:19 PM
>> I believe the SDP you must use is in the INVITE,
>> NOT in the 183 response.
>> Remember, when a SIP box sends SDP paramters,
>> it is saying "send me RTP to the RTP port described
>> in my SDP".
OK, my INVITE sends SDP but I need some information contained in response SDP
(media type for example) to open the RTP.
>> So, if a SIP box sends an INVITE with no SDP, then
>> the remote SIP endpoint would not be able to send
>> media to the SIP box because it wouldn't know where
>> to send it to.
>> All you have to do is make sure that the sender of
>> the INVITE has opened his RTP port for listening.
When I send the INVITE I open the RTP only after I have received a 200 OK or a
183. When I receive a 183, I open RTP only in reception and UserA can listen the
"message" received waiting the call go up.
But if it received 3 RTP (suppose they use tha same media type) which of the 3
"messages" the UserA have to listen?
Which of the 3 RTP stream I have to use?
>> Let me know if you agree.
>> Cheers,
>> Attila
Thanks
Lorenzo
P.S. Excuse me for my bad english!!
Attila Sipos
Software Engineer
DDI ( +44 1344 784918 )
<mailto:attilas@vegastream.com>
<http://www.vegastream.com>
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VegaStream : A World of difference for your Integrated Communications
> -----Original Message-----
> From: Lorenzo_Boffelli@alliedtelesyn.com
> [mailto:Lorenzo_Boffelli@alliedtelesyn.com]
> Sent: 20 February 2002 19:58
> To: sip-implementors@cs.columbia.edu
> Subject: [Sip-implementors] 183 response
>
>
>
> Hi,
> Suppose to use a SIP Proxy Server that make a call forking.
>
> UserA call UserB and Proxy forks the call to UserB1, UserB2 and UserB3
> Suppose UserB1, UserB2 and UserB3 respond with a 183 code with SDP.
> UserA receive tree 183 message with 3 different SDP and it
> has to open the RTP
> channel. Which SDP it must use?
>
> Thanks
> Lorenzo
>
>
> ___________________________________________
>
> Lorenzo Boffelli
> STRE Engineer
>
> Allied Telesis K.K.
> Head Office / 4F TOC Bldg, 7-22-17 Nishi-Gotanda,
> Shinagawa-ku, Tokyo Japan, 141-8635
> European R&D Center
> Piazza Tirana, 24/4 b Phone: +39 02 41411201
> 20147 Milano Fax: +39 02 41411260
> ITALY
> Email: lorenzo_boffelli@alliedtelesyn.com
> ___________________________________________
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
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