Please note that each UA will have preference setup for the codec
supported in the system, based on other parameters like bandwidth
utilization policy, quality of media, availability of DSP resources for
the particular codec type, system traffic..etc.
All these will govern such decision.
Regards
S
PRACK is considered as NON Invite Transaction, hence rules applicable to
NON Invite transaction should apply.
Refer to section 17.2.2 for all timer related query for the response.
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-imple
[mailto:nauman762-h...@yahoo.co.uk]
Sent: Wednesday, June 29, 2011 1:55 PM
To: sip-implementors@lists.cs.columbia.edu; Kumar Verma, Sunil (Sunil)
Subject: RE: [Sip-implementors] 18x response after OA complete?
Thank you. However can you confirn no more OA cycles can take place
using the 18x and PRRACK response
Yes, it is allowed.
If you see proxy and other entity, they need 18x response to retain the
transaction(Timer C).
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Nauman Sulaima
Hi,
I am not sure sending INVITE without SDP indicates call hold.
Can you please refer to which section in 3261 refer the same?
Invite without SDP can be considered as Request offer, and some time in
these cases far end responds with Complete codec list.
For call hold we need to either send INVITE
Hi Chandan,
I think the call flow you have mentioned has few things unclear..
VOIP 1 is at fault by not sending ACK to 487.
If Cancel is retransmitted then it seems 200 OK for cancel is not processed by
VOIP 1. This might be reason for missing ACK.
But if Cancel is retransmitted and for every
I think he is refereeing to SIP client, it can be both UAC/UAS.
Correct me if I am wrong.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Paul Kyzivat
Sent: Thursday, March 31, 2011 5:30 PM
To: sip
In case the request sent by user is forked by proxy and received back by
originator of request.
There will be use cases when we use call handover features.
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.c
Hi Tarun,
Port=0 is rejection of stream.
Now if only one stream is offered and it is rejected then UAC can
terminate the call.
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Answers inline:
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Tarun2 Gupta
Sent: Thursday, November 25, 2010 4:45 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] Query r
Hi,
Asymmetric payload is allowed in sip.
As SDP is receive capability hence:
A-> B (RFC2833 Payload :101)
B-> A (RFC2833 Payload :96)
Also based on what is negotiated the streaming will be using that
payload.
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@lists.
Hi,
The first Notify due to implicit subscription will contain the expire
time value for this subscription. If Subscription refresh request for is
not sent before that Subscription will be considered terminated.
Regards
Sunil Verma
-Original Message-
From: sip-implementors-boun...@list
Hi,
The purpose of Subscription is to notify the Referring party about the
state of the new call due to refer. This is mainly used to identify if
call is successful or failed.
Now based on response the referring party decided to terminates the call
or retrieve the referred call.
If before the co
Hi,
Is the call from UAC to UAS is via any proxy?
Ideally proxy will be running Timer C which will reset to new value with
18x
response, but I think it will not be more that 3 minutes. So Proxy
should have dropped both leg of the call.
Other way would be to use the Expire header in SIP Invite w
Hi Brandon,
I think if UA1 is aware of both the call legs then it will be good to
send Refered by as "us...@example.com". But please note that this is not
mandatory. As there could be many other mechanism to identify and
Authenticate the Referring party.
"Referred-By" is one of the mechanism to A
Hi Vivek,
As you may have proxy in the network which would have responded back
with 100 Trying and in turn have started timer C. Proxy will terminate
both legs of the call, if there is no response to the request it has
forwarded before timer C expire.
Also at UAC level you can have expire header
Hi Abhishek,
Can you elaborate more for three way handshakes and which part of three
way handshakes you think is not required?
100 Trying can be generated even for Re-INVITE if the processing of the
same is going to take more than 200ms. I don't think there is any such
restriction in RFC for this
Slight modification as I mentioned "sendrecv" at the at line instead if
"receive-only".
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Verma Sunil
Sent: Wednesday, September
Hi,
As per offer answer RFC for media attribute "send-only" response can be
"receive-only" or "inactive".
The choice is at the application to decide what needs to be done.
>From the call scenario I think response should be "inactive" as BOB has
put Alice on hold and has not yet retrieve the call.
Hi Anand,
I agree with Paul.
I am not sure why you have to put SDP and not m-line for Audio?
Do you want to share other media attributes and not Audio?
As per spec there could be many ways to initiate a call with hold
attributes.
You can use the rules mentioned below by Paul in offer as well.
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