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PUNJABI BOY | 1 May 2012 14:37
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Termination is Not replying to INVITE at ALL

Dear All,

I need the help from in one of my sip call scenario. Call flow is as below.

UA -> 187.28.162.5 -> SBC Realm 1 (200.182.99.75) -> Machine SIP Realm 1
(187.60.58.133) -> SBC Realm 2 (187.60.52.80) -> Termination Provider Realm
1 (200.216.239.196).

However the problem i am seeing rightnow is that whenever my SBC is sending
the call from from *second last entity to Termination Provider REALM1 then
the termination provider realm is not replying to the INITIAL invite at all.
*

Although i checked the content type value in message body and seeing
something odd, not sure if this is the valid headers or not as below.

UA--->to SBC(And in this SBC recieved the INVITE from UAC and forwarded to
next hope(Which means SBC Ingress realm and it sends the 100 Trying)

=====

Content-Length: 455

Content-Type: multipart/mixed;boundary=ssboundary

--ssboundary

Content-Length: 249

Content-Type: application/sdp
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PUNJABI BOY | 1 May 2012 20:02
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Re: Termination is Not replying to INVITE at ALL

Dear All,

Could anyone please help me out on the below problem please.

Thanks,
Nitin

On Tue, May 1, 2012 at 8:37 AM, PUNJABI BOY <nitinkapoorr <at> gmail.com> wrote:

> Dear All,
>
> I need the help from in one of my sip call scenario. Call flow is as below.
>
> UA -> 187.28.162.5 -> SBC Realm 1 (200.182.99.75) -> Machine SIP Realm 1
> (187.60.58.133) -> SBC Realm 2 (187.60.52.80) -> Termination Provider Realm
> 1 (200.216.239.196).
>
>
> However the problem i am seeing rightnow is that whenever my SBC is
> sending the call from from *second last entity to Termination Provider
> REALM1 then the termination provider realm is not replying to the INITIAL
> invite at all.*
>
>
> Although i checked the content type value in message body and seeing
> something odd, not sure if this is the valid headers or not as below.
>
>
> UA--->to SBC(And in this SBC recieved the INVITE from UAC and forwarded to
> next hope(Which means SBC Ingress realm and it sends the 100 Trying)
(Continue reading)

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Shanbhag, Somesh (NSN - IN/Bangalore | 2 May 2012 09:57

Re: Termination is Not replying to INVITE at ALL

Hi Nitin,

There could be multiple reasons on why the terminating UA (or UAS) is
behaving without sending the response.
The SDP looks okay. I would say may be its not liking multipart content
type or its not able to decode application/isup.

But typically, all TU have to be designed in a way that it will receive
the messages liberally and deliver strictly (aka RFC 3261).
I would recommend to take trace/logs in the terminating UA to see what's
going wrong. In either way, it should have sent negative response (400
etc).

Regards,
Somesh

-----Original Message-----
From: sip-implementors-bounces <at> lists.cs.columbia.edu
[mailto:sip-implementors-bounces <at> lists.cs.columbia.edu] On Behalf Of ext
PUNJABI BOY
Sent: Tuesday, May 01, 2012 8:03 PM
To: sip-implementors <at> lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Termination is Not replying to INVITE at
ALL

Dear All,

Could anyone please help me out on the below problem please.

Thanks,
(Continue reading)

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Naarumanchi Kaushik | 4 May 2012 11:16
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SIP Servers that support RFC 5626?

Can anyone suggest SIP servers that support RFC 5626?

Thanks
Kaushik.
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Jean Deruelle | 7 May 2012 11:33
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Re: SIP Servers that support RFC 5626?

We support parts of it in Mobicents Sip Servlets
http://code.google.com/p/sipservlets/wiki/Welcome?tm=6

Jean

On Fri, May 4, 2012 at 11:16 AM, Naarumanchi Kaushik <
shankarkaushik <at> gmail.com> wrote:

> Can anyone suggest SIP servers that support RFC 5626?
>
> Thanks
> Kaushik.
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors <at> lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
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pranab sahoo | 7 May 2012 14:01
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issue in sip

Hi all,

          Hope you all doing well.

I have a doubt regarding cseq.
suppose I made a complete audio call between user A and B without
disconnecting call.In this case cseq is 1 INVITE.
In middle A want to send video from one way i.e from A to B.so in this case
cseq value will be 2 INVITE or 1 INVITE.
Again after one way video conversion between A to B,B also wants to make
video call from B to A.so what will be the value of Cseq in this case?
Again call id value will change or not?

waiting for quick reply.
Thanks in advance.

pranab
Brazil
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Shanbhag, Somesh (NSN - IN/Bangalore | 7 May 2012 14:06

Re: issue in sip

Hi Pranab,

(1) Re-INVITE from A to B: CSeq: 2 INVITE
(2) Re-INVITE from B to A: CSeq: 2 INVITE
(3) Call-ID shall remain same until BYE

Best Regards,
Somesh

-----Original Message-----
From: sip-implementors-bounces <at> lists.cs.columbia.edu
[mailto:sip-implementors-bounces <at> lists.cs.columbia.edu] On Behalf Of ext
pranab sahoo
Sent: Monday, May 07, 2012 2:02 PM
To: sip-implementors <at> lists.cs.columbia.edu
Subject: [Sip-implementors] issue in sip

Hi all,

          Hope you all doing well.

I have a doubt regarding cseq.
suppose I made a complete audio call between user A and B without
disconnecting call.In this case cseq is 1 INVITE.
In middle A want to send video from one way i.e from A to B.so in this
case
cseq value will be 2 INVITE or 1 INVITE.
Again after one way video conversion between A to B,B also wants to make
video call from B to A.so what will be the value of Cseq in this case?
Again call id value will change or not?
(Continue reading)

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Alok 2 Tiwari | 7 May 2012 14:10
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Re: issue in sip

Hi Pranab,

As per RFC-3261, section 12.2.1.1,

If the local sequence number is not empty, the value of the local
   sequence number MUST be incremented by one, and this value MUST be
   placed into the CSeq header field.  If the local sequence number is
   empty, an initial value MUST be chosen using the guidelines of
   Section 8.1.1.5.  The method field in the CSeq header field value
   MUST match the method of the request.

In your scenario,

 - A call is connected from A (originating party) to B (Terminating party), CSeq will be 1.
 - when A want to send video from one way i.e from A to B, the CSeq will be incremented and it will be 2.
 - When one way video conversion between A to B,B also wants to make video call from B to A, CSeq can be any
arbitrary number.

Also, Call-ID remains unique throughout the dialog and it cannot be changed.

Regards,
Alok Tiwari
Aricent

-----Original Message-----
From: sip-implementors-bounces <at> lists.cs.columbia.edu
[mailto:sip-implementors-bounces <at> lists.cs.columbia.edu] On Behalf Of pranab sahoo
Sent: Monday, May 07, 2012 5:32 PM
To: sip-implementors <at> lists.cs.columbia.edu
Subject: [Sip-implementors] issue in sip
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Honsha N Kalita | 10 May 2012 11:24
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SIP ALG with SIP Conntrack

Hi,
If anybody has used SIP connection tracking/NAT netfilter modules
["nf_conntrack_sip" and "nf_nat_sip"] and successfully mangled [
NAPT-ed] SIP messages from user-space , could you please share some
details ?

 Can it be done via IPTABLES with some options itself,  or we need
kernel modules to be written using APIs of these modules ?

Regards
Honsha N K
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sunil sinha | 10 May 2012 12:39
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[RFC 3261] Priority header value

Hi All,

The header field can have the values "non-urgent", "normal", "urgent", and
"emergency". How do we define these header field values with our practical
life scenario?

Thanks & Regards

-sunil
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