Re: [Sip-implementors] Send media in video call Vs audio call in SDPAnswer/Offer model

2008-12-15 Thread Raghavendra Kamath
Hi Richard, Why does A require the 200 OK to be able to decode the packets? The 200 OK contains information about what B wishes to receive. A should already have opened its decoders as soon as it send the INVITE outwards. It knows the PayloadTypes that B is going to send it. So it can also detect

Re: [Sip-implementors] two-way hold/resume

2008-12-03 Thread Raghavendra Kamath
Responses inlined.. -Original Message- From: Paul Kyzivat [mailto:[EMAIL PROTECTED] Sent: Thursday, December 04, 2008 11:06 AM To: Raghavendra Kamath Cc: sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implementors] two-way hold/resume Raghavendra Kamath wrote: > P

Re: [Sip-implementors] two-way hold/resume

2008-12-03 Thread Raghavendra Kamath
Paul/Kaiduan, It was pretty insightful. So, applying Paul's logic, is the following a valid scenario? A,B are video endpoints and in a call with 2-way audio video flowing. Videocalls support the concept of presentation, wherein only one participant is allowed to generate videocontent for all to

[Sip-implementors] SIP and BFCP

2008-11-05 Thread Raghavendra Kamath
Hi All, We are in the process of implementing BFCP for the purpose of floor control in video presentation. The draft that defines this is RFC 4582. But I see that this spec requires exchange of BFCP protocol messages over TCP. I would like to know from anyone about what is the feasibility of this

[Sip-implementors] Session Timer and NAT

2007-10-22 Thread Raghavendra Kamath
Hi All, Session Timers seem to be one way to handle the NAT issues. I wanted to know if this mechanism also helps in keeping the bindings for RTP/RTCP alive apart from the signalling port itself? If so then how is it implemented at the NAT/ALG? Is it so that as soon as the reINVITE is received th

[Sip-implementors] Bandwidth modifiers in SIP

2007-09-26 Thread Raghavendra Kamath
Hi, I am sure this has been discussed earlier, please point me to the thread if so. My concern is related to the bandwidth modifiers that are used in many SIP implementations that support video. We have CT,AS,TIAS. Now AS,TIAS are pretty much well defined. These can be added either at the session