Newbie Question - B End response when TO Party is a real extension/device

[Craig]

Hi

Apologies for this is a very basic question but I have read through the tutorials and I cannot see my error. I am very new to this excellent application, so it is clear I am doing something fundamentally wrong.

I have managed to write scenarios that can register SIPinspector UA, make calls and introduce RTP via my companies Hosted Telephony cloud.

But all these scenarios usually require a human to answer the dialled (TO) party which is typically a real physical handset with associated extension number.

What I have been trying to simulate is 486 or 603 responses to invites from SIPinspector.

So I setup SIPinspector to register the FROM party and invite out to the TO party (which is a real extension within my Hosted Telephony environment).

However when my scenario sends back a 486 message from the TO party the device actually rings out and the 486 is ignored.

Now I guess this is because the invite has already hit the TO party and the 486 is never acted upon as I can see in the Scenario Screen on run the 486 message EXP = 0 and instead a UNEXP = 180 for ringing.

---> REGISTER	1	0
<--- 100*	0	0
<--- 407*	1	0
---> REGISTER*	1	0
<--- 200	1	0
---> INVITE	1	0
<--- 100*	1	0
<--- 486	0	3	1x180,1x200,1xBYE
---> ACK	0	0

What I am trying to convey is that I am struggling with the concept of how SIPinspector handles both the TO and FROM parties call flow, when the TO party is a real device/extension.

I apologise for the basic nature of this question, but I think once I get my head around this particular aspect, SIPinspector will become an invaluable tool for my work.

I have included my basic Scenario text for whoever chooses to help me out.

Thank you in advance.

Craig

---------->

REGISTER sip:[remote_ip]:[remote_port] SIP/2.0

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

From: <sip:[fromUsername]@[remote_ip]:[remote_port]>;tag=24de4a5dbe4ef

To: <sip:[fromUsername]@[remote_ip]:[remote_port]>

Call-ID: [call_number]@[local_ip]

CSeq: [cseq] REGISTER

Max-Forwards: 70

Expires: 3600

Contact: <sip:[fromUsername]@[local_ip]:[local_port];transport=[transport]>

User-Agent: SIPInspector_v_[ver]

Content-Length: 0

<----------*

100 Trying

<----------*

407 Proxy Authorization Required

---------->*

REGISTER sip:[remote_ip] SIP/2.0

From: <sip:[fromUsername]@[remote_ip]>;tag=24de4a5dbe4ef

To: <sip:[fromUsername]@[remote_ip]>

Call-ID: [call_number]@[local_ip]

CSeq: [cseq+1] REGISTER

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

Contact: <sip:[fromUsername]@[local_ip]:[local_port];transport=[transport]>

User-Agent: SIPInspector_v_[ver]

Accept-Language: en

Max-Forwards: 69

P-Asserted-Identity: <sip:[fromUsername]@[remote_ip]>

Proxy-Authorization: [authentication username=[authUsername]; password=[authPassword];]

Expires: 3600

Content-Length: 0

<----------

200 OK

---------->

INVITE sip:[toUsername]@[remote_ip]:[remote_port];transport=[transport] SIP/2.0

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

From: <sip:[fromUsername]@[remote_ip]:[remote_port]>;tag=452352542352354325

To: <sip:[toUsername]@[remote_ip]:[remote_port]>

Call-ID: [call_number]@[local_ip]

CSeq: [cseq+1] INVITE

Max-Forwards: 70

Contact: <sip:[fromUsername]@[local_ip]:[local_port];transport=[transport]>

User-Agent: SIPInspector_v_[ver]

Content-Type: application/sdp

Content-Length: [len]

v=0

o=111 843670094 843670094 IN [IP_ver] [local_ip_sdp]

s=-

c=IN [IP_ver] [local_ip_sdp]

t=0 0

a=sendrecv

m=audio [media_port] RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

<----------*

100 Trying

<----------

486 Busy Here

---------->

ACK sip:[toUsername]@[remote_ip]:[remote_port];transport=[transport] SIP/2.0

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

From: <sip:[fromUsername]@[remote_ip]:[remote_port]>;tag=452352542352354325

To: <sip:[toUsername]@[remote_ip]:[remote_port]>;[peer_tag_param]

Call-ID: [call_number]@[local_ip]

CSeq: [cseq] ACK

Max-Forwards: 70

Contact: <sip:[fromUsername]@[local_ip]:[local_port];transport=[transport]>

User-Agent: SIPInspector_v_[ver]

Content-Length: 0

[Zarko C]

Hi Craig!

Your scenario in its current form does following: registers and makes a call to some other device. The scenario also says it expects to get 100 Trying and then 486 Busy Here. Instead, the hosted telephony application sends 180 Ringing.

What I think you are trying to simulate is to to have parties A and B. Both A and B are 'real' and configured on the hosted telephony application. A then calls B and B either automatically answers the call or rejects it with some error code. To achieve that have two instances of SIP Inspector running on the same PC (each bound to the same IP but different SIP signalling and media port). One will initiate the call and the other will receive and either answer or reject it. So respective scenarios would go something like this:

Client A	Client B
---> REGISTER	---> REGISTER
<--- 100*	<--- 100*
<--- 407*	<--- 407*
---> REGISTER*	---> REGISTER*
<--- 200	<--- 200
---> INVITE	New Dialog
<--- 100*	<--- INVITE
<--- 486	---> 100
---> ACK	---> 486
	<--- ACK ...

NOTE:To simulate Client B behavior take a look in scenario directory and 'Client_reg_incoming_call.txt' in particular. And in it modify responses it sends back. Instead of 180 and 200 put 486 or 603.

All the signalling will go through the hosted telephony app and SIP Inspector will emulate SIP endpoints.

I hope this helps.

Regards,

Zarko

[Craig]

Thank you very much Zarko for your fast and excellently detailed response.

I literally just thought about having two instances open and was trying to get my head around it. Your response should hopefully ensure I get to my desired objective a lot sooner.

Thank you again.

Craig

[Craig]

UPDATE

THANK YOU !!!!!!!!!!!!!!!!!!!!!!

All sorted an running like a dream. Will try and build on this knowledge.......

On Monday, 5 September 2016 14:35:12 UTC+1, SIP Inspector wrote:

[Zarko C]

:-)

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