Hi Josh,
One last followup: I was curious to see exactly what happens when
dhclient renews my IP address, so I captured it with wireshark and found
that in the DHCP request, dhclient specifies the IP address it is
requesting to be renewed (the same IP that was originally leased). The
DHCP serv
Hi Josh,
What leads you to believe renewing a DHCP lease will drop your call?
Think about this: have you ever had an HTTP download break because your
computer needed to renew it's DHCP lease? I've never experienced such a
thing.
I'm not an expert by any means with DHCP, but if your computer se
Mayank and friends,
I'm trying to authenticate a statement by a voip/sip system integrator that
dhcp has stability and reliability issues because when the phone rings
(every time, according to this company) it has to check its lease for
renewal and potentially renew the lease before the call can c
Hello,
I have a question regarding the Digest-URI used to calculate the response
parameter which is put into an Authorization header. From RFC 3665 it seems
to me that the Digest-URI is the same as the Request-URI. However, in
draft-smit-sip-auth-examples the From-URI is clearly used in all
calcul
On Thu, Mar 31, 2011 at 5:21 PM, isshed wrote:
> yes Paul, you get the question right?
> do you think a client can send the 482...by client i mean a sip endpoint..
>
We do that in the python-sipsimple library in the following case: user
has added 2 accounts, and the library will use same local Co
yes Paul, you get the question right?
do you think a client can send the 482...by client i mean a sip endpoint..
Thanks
On Thu, Mar 31, 2011 at 5:29 PM, Paul Kyzivat wrote:
> Is this a trick question?
>
> A *client* never sends responses. The thing that sends (any) response is
> an server.
>
> I
31 mar 2011 kl. 15.42 skrev Worley, Dale R (Dale):
>
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Nitin Kapoor
> [nitinkapo...@gmail.com]
>
> I am facing the issue with one of my
From: sip-implementors-boun...@lists.cs.columbia.edu
[sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Nitin Kapoor
[nitinkapo...@gmail.com]
I am facing the issue with one of my client, where my Termination is sending
183 with SDP but my UAC i
On 3/31/2011 9:04 AM, Josh Roberts wrote:
> Resending, and hoping for a response. To avoid spamming, I won't resend
> again. Please advise if I should be posting to a different list.
>
> On Sat, Mar 26, 2011 at 15:00, Josh Roberts wrote:
>
>> Anybody have:
>>
>>
>> 1. any rfc reference for b
Hi Nitin,
You can capture the RTP near the UAC and see whether the media is coming
up to the UAC after 183. If media is coming to the UAC, there might be a
setting in the UAC to enable playing RTP after receiving 183. I have
seen such configurations in some of the end points
Hope this helps
BR,
On 3/31/2011 8:56 AM, pranab sahoo wrote:
> Hi All
> Thanks all of you providing such a platform to clarify SIP/IMS doubts
>
> why we use "P-Visited-Network-ID header" in IMS?
> What are the advantages of using this?
> which UE or proxies are going to use it?
> If we will not use this p-header what
On 03/31/2011 04:13 PM, Nitin Kapoor wrote:
> Dear All,
>
> I am facing the issue with one of my client, where my Termination is sending
> 183 with SDP but my UAC is unable to hear any destination country ringback.
>
>
Hello,
Although this is very weird, but does the SBC also send a 180 Ringin
Dear All,
I am facing the issue with one of my client, where my Termination is sending
183 with SDP but my UAC is unable to hear any destination country ringback.
However after connecting the calls there is bi-directional media between 2
end users. I never faced this kind of issue earlier. Could
Resending, and hoping for a response. To avoid spamming, I won't resend
again. Please advise if I should be posting to a different list.
On Sat, Mar 26, 2011 at 15:00, Josh Roberts wrote:
> Anybody have:
>
>
>1. any rfc reference for best practice on voip deployment regarding
>whether to
Hi All
Thanks all of you providing such a platform to clarify SIP/IMS doubts
why we use "P-Visited-Network-ID header" in IMS?
What are the advantages of using this?
which UE or proxies are going to use it?
If we will not use this p-header what are difficulties we will face.
thanks inadvance
Prana
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
See RFC 3398 section 7.3.
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-
> implementors-boun...@lists.cs.columbia.edu] On Behalf Of Md Faruk Apel
> Chowdhury
> Sent: Thursday, March 31, 2011 2:56 AM
> To: sip-implementors@lists.cs.columbia.edu
> Su
Is this a trick question?
A *client* never sends responses. The thing that sends (any) response is
an server.
I guess the question is whether there is any case where a UA can send a 482?
Thanks,
Paul
On 3/30/2011 10:57 AM, Brett Tate wrote:
>> Is there any scenario or use case
In such a case SIP client should send ACK to MGC to complete the INVITE request
for starting Media flow or Not.
From: ashok kumar [mailto:ash@gmail.com]
Sent: Thursday, March 31, 2011 11:02 AM
To: Md Faruk Apel Chowdhury
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implement
There is no equivalent message for ACK in ISUP and the call gets established
after ANM on the PSTN side.
On Thu, Mar 31, 2011 at 12:26 PM, Md Faruk Apel Chowdhury <
mdchowdh...@etisalat.ae> wrote:
>
>
> Call from SIP to PSTN
>
>
>
> SIP client-MGC-PSTN
>
>
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