Hello Chandan,
You can check out Codenomicon's SIP server and client side negative
testing tools or also Ixia's IxDefend. These are commercial tools but
worth taking a look at. Also PROTOS SIP Test Suite which is a little out
of date but still good for negative testing.
http://www.ee.oulu.fi/re
Hi Chandan
I really do not think you can get a complete VoIP test case suite? Better to
develop it urself
I suggest you can do following: If your VoIP product is based on SIP, then list
down all the RFCs or I-Ds it supports. Then based on that write test cases for
every requirement / artifact
I would suspect the packetization interval in the answer; may be the offerer
did not like this.
a=ptime:40
--
Regards
Harsha
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2009/5/22 :
> Guys,
>
> I am looking more to a codec mismatch we are looking into this at present.
> we send an invite with all codec options then the recipient replies with
> yes and use codec G729r8 I think we maybe trying to communicate in G711.
> Does this sound plausable? Bare with me on th
Guys,
I am looking more to a codec mismatch we are looking into this at present.
we send an invite with all codec options then the recipient replies with
yes and use codec G729r8 I think we maybe trying to communicate in G711.
Does this sound plausable? Bare with me on this 1 as I am very new t
What "complete VoIP product?"
What VoIP technology stack? Do you want an H.323 test suite?
Complete description FAIL. Question cannot be answered as posed.
chandan kumar wrote:
> Hi ,
>
> Please coud any one help me in finding VOIP test cases Negative & positive
> too .This for testing the
2009/5/22 chandan kumar :
> 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?
Must it be so free? if you are going to sell a product and earn money
with it, why
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.
Bollywood news, movie reviews, film trailers and more! Go t
My guess is that something it didn't like in SDP.
[offer]
v=0
o=- 3800 3800 IN IP4 10.99.1.1
s=-
c=IN IP4 10.99.1.1
t=0 0
m=audio 64580 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
[answer]
v=0
o=iS3000 1 1 IN IP4 10.99.11.1
s=-
Sorry Guy's
Hopefully the trace below is more informative.
[1 bytes missing in capture file]INVITE
sip:4...@10.99.11.1:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.99.1.3:5060;branch=z9hG4bK2604850560-60281937
Max-Forwards: 70
Allow: INVITE, BYE, CANCEL, ACK, INFO, PRACK, OPTIONS, SUBSCRIBE, NO
2009/5/22 :
> Hi All,
>
> Could someone take a look at my trace below and advise where I am
> receiving a bye just as the call has established?
It's obvious that in your trace messages from UAC (INVITE, BYE...) are
hidden. Could you try to improve your problem description by showing
useful info?
I am not sure of the question you are asking.
Looks like the capture has all the messages from UAS and there are no messages
from UAC.
BYE is being sent from UAC and 200 OK for BYE is received ( last message ).
If you are asking for the reason for BYE, it would be helpful if you can paste
the
Hi All,
Could someone take a look at my trace below and advise where I am
receiving a bye just as the call has established?
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.99.1.3:5060;branch=z9hG4bK2604850560-60281937
From: "M.Wheldon" ;tag=0_2604850560-60281938
To: ;tag=483089415066
Call-ID: 2604850560-6
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