Hi all,
I have tried to reproduce the issue using Sipp tool. I have added extra
paramets of rport=5060 in Via Header . made call between Voip1 & Sipp tool .
For the Cancel sequence my Voip responds with ACK.
I cannot point its an issue with Voip-1 ,because it works with toher servers
respo
Hi all,
Thanks a lot for your valuable answers
The 2 Voip phones what ever i have used for this scenario works perfectly for
the Cancel sequence with other servers like trixbox,seimens,Teksip etc.
Voip1 responds perfectly for 487 by sending ACK & there are no retransmissions
of Cancel.
A
Hi ,
Iam facing an issue with Cancel. Please find the Cancel.txt which has the
call cancel sequence..
Voip-1 makes a call throough Server ,before Voip2 accepts it cancels call.
1)Voip-1 sends Cancel ,Server responds with 200 Ok.
2)Server sends 487 Request Termination ,But Voip1 is
Hi ,
small Query regarding a CANCEL request. Say make call from UA to UAS via
proxy(B2BUA) ,before UAS accepts the call .
1.UA cancels the request. proxy responds with 200 Ok .
2.Proxy sends CANCEL to UAS & uAS respond with 200 ok.
3.UAS sends 487 to proxy & proxy send ACK .
4.But proxy se
Hi,
Its a samll query regarding headers. Priority & subjects headers can be sent
seperately in INVITE (or) both must be sent together in INVITE.
Please let me know the rfc no to go through different headers in detail.
Best Regards,
CKumar.
___
Si
Hi,
Please can some one clarify the below queries regarding Session timer.
A UAC starts sending an INVITE. This includes a Supported header field
with the option tag 'timer', This INVITE passes through a proxy & this proxy
inserts Min-SE value of 1800 .
1. Finally the request arrives a
Hi,
if we want to call a analog line on a PBX which does not act as a
SIP-Server.Is there a way to do this?
The call by IP INVITE look like "INVITE sip:192.168.10.234...".
If we want to call a analog phone on a Peer to Peer PBX with a invite like
"INVITE sip:9411@192.168.10.7..." can it
l Kyzivat
Subject: Re: [Sip-implementors] Query on SDP hold
To: sip-implementors@lists.cs.columbia.edu
Date: Friday, 21 January, 2011, 3:06 PM
On 1/19/2011 1:49 PM, Chandan Kumar wrote:
> Hi,
>
> As per O/A model rfc 3264 re-invite offer with SDP containing connection
>IP 0.0.0
Hi,
As per O/A model rfc 3264 re-invite offer with SDP containing connection IP
0.0.0.0 is not recommended . Since UA neither send any RTP/RTCP as the
destination media address is not known nor receive any RTP/RTCP from 0.0.0.0.
we are facing interop issue with B2BUA when a re-Invite off
>
>>
>> How to deal with this issue& which is correct option .
>>
>>
>> Best Regards,
>> Chandan.
>>
>> --- On Tue, 4/1/11, Harlin Dyvia Helina Sathianathan<
>> harlin.sathianat...@aricent.com> wrote:
>>
>>
>> From: Harl
responds with a=sendrecv ?
How to deal with this issue & which is correct option .
Best Regards,
Chandan.
--- On Tue, 4/1/11, Harlin Dyvia Helina Sathianathan
wrote:
From: Harlin Dyvia Helina Sathianathan
Subject: RE: [Sip-implementors] SIP Call/Hold Scenario
To: "Chandan Kumar
Hi All ,
This is an query regarding call hold scenario .
Two phones A,B are registered to SIP proxy. Let's say A & B are in call.
1.Now A holds the call by changing medi attribute in SDP to a= sendonly.
2.Proxy sends signalling message Trying.
3..Proxy sends 200 OK with SDP to Phone A in
Hi,
Iam Pretty new to this SIP ,I have small query on SIPp script.
My Voipohne is failing when it recieves a SIP INIVTE with P-asserted Identity.I
have done some modifications to my code.
Generally P-asserted Identity is provided by providers.
To simulate this issue I wrote a Sipp script
Hi ,
. According to my understanding P-asserted Identity is added by Proxy servers
within in the trusted domains . Please correct me if Iam wrong.
one of our customers claims if P-asserted Identity is there in SIP header
,Incomming call is failing.Is any one aware of adding this feature in a
Hi ,
I have implimented SIP ALert-Info header in INVITE message .But I would like to
know whether its valid for sending Aler-Info header in final Responses.
If any one know how to test this feature using Astersik trix box?Please let me
know
Best Regards,
CKumar.
__
Hi all,
I need a suggestion regarding the work Iam doing & how to improvise it.
Actually Iam working as IP-phone product tester ,Role of my work is to test
complete phone (includes SIP features,Audio,Video etc)testing on some
different routers ,servers etc .ALl the things are done manua
Hi all,
Iam pretty new to thi SIP field, so please give an on below questions.
Could anyone give me difference between STUN & UPNP? Are these two are
dependant on each other.
Any VOIP based phones should have both the functionalities?
How can we identify the type of router mean Static NAT
Hi all,
Could any clarify what does HOP do in SIP?
whether HOP is related to a register request or Call request?
Issue faced : Phone comes up & tries to regsiter to the server.Lets say the
server is not available.So registration fails .Now I will try to make an IP
call to other SIP phone.Ca
Hi All,
Iam very much confused about sip expire time (during registration).
Say UserAgent registered to the Proxy . I want to know how long the UA will be
in the Proxy,When the UA agent again will send the Register request.
Ex:In my Code say Iam sending a register request after every 5 min (
Hi ,
Please coud any one help me in finding VOIP test cases Negative & positive
too .This for testing the complete VOIP Product.
Any freely available that helps for my testing?
Thanks in advance
Regards,
chandan.
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Hi ,
Iam testing IP phone which supports video & Audio.Iam facing an issue like
.Iam testing on 2 DSL lines . So End users are on different NAT's. I have
registered both the users to Public SIP servers( freel available servers for IP
calls , using SIPgate).Registration happens.I made call ,ca
Hi ,
Iam testing a full fledge voice & video over IP phone using SIpgate.
Iam facing some issues,Could any one explain the reason.
1.Two end users are registered to the SIPgate server.say A(India) & B(USA)
A could not call B, it tries to reach B ,request time out (408 response).At
same scenario
Hi ,
Are there any free tools available to test SIp functionality & also want to
test with diferent SIP servers.
I want to test a developed Voip phone while testing i want to know more about
the SIP responses & messages, real time problems etc
Please help me to find any tools and freely
Hi ,
Could any one let me know whats is the difference between outbound proxy
,registrar server & register server.
cheers
chandan.
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