Thank you Paul and Vivek.
Paul,in line some clarification on question 3. 

     On Monday, October 19, 2015 7:07 AM, Paul Kyzivat <pkyzi...@alum.mit.edu> 
wrote:
   

 I'm going to differ with Vivek to some extent. See inline below.

In any case, I suggest reading RFC6337.

On 10/19/15 2:21 AM, Talwar, Vivek (Nokia - IN/Noida) wrote:
> Hi Sipuser,
>
>    Please find the responses as per scenario:
>
> 1. Its valid as terminating endpoint is mentioning its preferences although 
> the call will be established on codec2.
> 2. Yes its valid. Scenario can be of some early media announcement. Initially 
> call was established on codec 2 but some how call was say rejected and some 
> announcement is now being played from network and Media Server in picture 
> will use codec4.
> 3. Similar to 2.
> 4. Yes its possible to as far as stream used is same. It stream is new then 
> new m =line needs to be added.
> 5. Didn't get this question.
>
> Thanks and Regards,
> Vivek Talwar
>
> -----Original Message-----
> From: sip-implementors-boun...@lists.cs.columbia.edu 
> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of EXT K 
> sipuser
> Sent: Monday, October 19, 2015 3:20 AM
> To: Sip-implementors
> Subject: [Sip-implementors] offer-questions.
>
> Hi,I have few questions.

The questions are hard to read. I'm reformatting them.

> 1) INVITE with codec1,codec2,codec3200ok with codec2, codec3.
> Is this ok to add codec3?

- INVITE with codec1,codec2,codec3
- 200ok with codec2, codec3.
- Is this ok to add codec3?

Yes it is. This means that either codec2 or codec3 may be used in the 
session, or a combination of the two.

A common example of this is with g711 + telephone-events.
But it is allowed with any combination of codecs.

This is becoming more important because now there is active use form 
multiple media streams within a single media session. These may use 
different codecs. But a single stream can also switch codecs.

Some implementations can't handle this. In that case they should 
negotiate down to a single codec, or a combination of codecs they can 
use concurrently. (E.g. telephone-events and one other.)

> 2)INVITE with codec1,codec2,codec3183 received with codec2.Update Received 
> with codec4In early dialog state is this valid?

- INVITE with codec1,codec2,codec3
- 183 received with codec2.
- Update *Received* with codec4
- In early dialog state is this valid?

You don't say that that the 183 is reliable (100rel). If it was then 
there would be a PRACK and 200 response to that.

The update is only valid if the 183 was reliable.

> 3)INVITE with codec1,codec2,codec3183 received with codec2.Update send with 
> codec4In early dialog state is this valid?

- INVITE with codec1,codec2,codec3
- 183 received with codec2.
- Update *send* with codec4
- In early dialog state is this valid?

How is this different from (2)? The only difference I see is *received* 
vs. *sent*. But every message received must have been sent.

Is (2) assuming the update is opposite direction to invite, while in (3) 
the update is in the *same* direction as the update?

In any case, my comments on (2) also apply to (3).
Adi>>I mean in both the case2 and case3 , 183 is send reliably >>if SUT does 
not like codec4, can it reject with 4xx response in early dialog state?>>will 
that end the call?>>can it continue with codec2 after sending a failure 
response?

> 4)INVITE with codec1,codec2,codec3200 OK received with codec2.--call is 
> connectedUpdate Received with codec4

- INVITE with codec1,codec2,codec3
- 200 OK received with codec2.
- call is connected
- Update Received with codec4

Yes, this is fine. But the recipient of the update can reject it and 
continue the call with codec2.

The sender of the update needs to be prepared to receive either codec2 
or codec4 until the response to the update is received. He should keep 
sending codec2 until he knows whether codec4 has been accepted.

> 5) if SDP offer has RR and RS=0.can answer be non-zero RR and RS?RR in offer 
> should be equal to RS in answer right?

I don't know what RR and RS mean in this context.In general, in SDP we have RR 
and RS.if RR and RS are 0 in offer, can the answer RR/RS be non-zero?is the 
following statement correct?the bandwidth in Offer RR should be equal to RS(or) 
less  in Answer and the bandwidth in Offer RS should be equal to RR(or) less  
in Answer ? .

> Can you please help in understanding these?

Good luck.

    Thanks,
    Paul

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