[SIPForum-discussion] No Ring Back Tone Issue

Anthony Orlando avorlando at yahoo.com
Thu Sep 4 08:38:26 UTC 2008


RFC3398 describes the inter-working yet it leaves out the sdp.  My experience is that we first receive 183sdp from early ACM then CPG which maps to 180 w/o sdp.  This casued our app server to provide ring tone as well as the inband tone from the pstn.

Our appserver has a flag for this behaviour which the 180w/o sdp will not trigger the ringtone to be provided by the media server.

Unfortunatley I have found that while RFC3398 and Q1912.5 describe the call flows well they don't include sdp.  This is an area that needs addressing.

--- On Thu, 9/4/08, Herve Jourdain <herve.jourdain at mstarsemi.com> wrote:
From: Herve Jourdain <herve.jourdain at mstarsemi.com>
Subject: Re: [SIPForum-discussion] No Ring Back Tone Issue
To: "'ramon nolasco'" <rpnolasco at yahoo.com>, discussion at sipforum.org
Date: Thursday, September 4, 2008, 1:56 AM




 


 







Hi, 

   

I can’t remember a « normative »
RFC that would state just that… 

As far as I know, SDP and
18x handling is implementation dependant, though a common use – that seems
widely accepted – would be to have the SDP in 183 for ring back tone,
when PSTN gateway is involved. 

But again, this doesn’t
seem to me to be “normative”. 

   

And interoperability
tests show that SDP can be sent both in 183 – most common case –
but also in 180 – happened with some switches… I even saw 183
responses without SDP… 

In the end, for the
playing of ring back tone, I decided to do it on both 180 and 183 messages,
provided they had SDP inside, and generate local tone if no SDP is provided. The
background for this being that if anyone cares enough to send some SDP, then it
should be worth “listening” to it. But again, this is an
implementation choice… 

   

So I’m not sure
there is a “definite normative” answer to your problem, it sounds
like one - of many - SIP interoperability issue. 

One of you or your partner
will have to take some steps there. Your current way is “legal”
regarding SIP and your partner could handle it and still be SIP compliant, and
if you modify your behaviour to conform to what your partner requires, you will
also still be SIP compliant. 

   

So to me, it sounds like
you “just” need to make a decision about which way you want to go
with your partner, and implement it: you’ll still be SIP compliant either
way. 

My 2 cents: SDP in 183, again,
seems to stick more to “common practices” in SIP, in my opinion. 

   

Hope this helps. 

   

Regards, 

   

Herve 

   









From:
discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of ramon nolasco

Sent: mercredi 3 septembre 2008
03:21

To: discussion at sipforum.org

Subject: Re: [SIPForum-discussion]
No Ring Back Tone Issue 



   





  









Dear VB,
Raghul, Cliff, Torstein, Vijay, Shyw13, Raghavendra and All,  

  

Thanks
and appreciate so much in sharing your know how, comments and tips, which
all are based on the existing standards and solid experience and
background. 

  

One most
important thing I need to clarify.... 

  

"As
per standard, should our 183 response to the callee, SHOULD already
have SDP? For them then to play the ring back tone locally? Though
we have the SDP on our 180 response, but they can't consider processing it
somehow"  

  

  

Below is
the email explanation of our partner on how their SIP system handles our 183
and 180 responses. And with this, they wanted us to embed SDP in our 183
message. Is their request to us still standard for SIP? 

  

SIP
Partner wrote: 

  

With
regarding on mentioned packet data SDP(Session Description Protocol RFC2327)
Yes, that should be the important keywords and necessary packet in the sip
connection for figure out this case . 

We
think following matter is the highlighted point which is complicating. 

------------------------------------------------------------------------ 

1) 

Our
source device can accept the alerting message "only one time". but
your destination device didn't have SDP packet data when you advertise us the
"FIRST"alerting message. 

That
is why we couldn't realize the ring tone , even though your destination device
prepared the SDP packet data on the second exchange of alerting.  

(Our
source device cannot accept second contact for alerting from destination
device.) 

2) 

So
,we 'd like you to figure out the cause of missed SDP data on your first
alerting message.  

We
believe so that our source device can realize the ring tone sounds,  

if
we receive your SDP on your first contact for Alerting message. 

------------------------------------------------------------------------ 

Point) 

"Alerting
message & SDP which you have prepared are correct", but our source
device need you to succeed it on the "the first time" of  

alerting
negotiation. 

  

Sorry
for my poor explanation , but i hope it can be your good help... 

  

Thank
you 



   





I wish that you all say and comment, for it's in the
standard, that SDP in 183 SHOULD NOT be the basis for playing the ring
back tone for it's the SDP in the 180 that SHOULD be the basis. 





  





Regards, 





Mon 







  





----- Original Message
----

From: Torstein Knutsen <torstein.knutsen at gmail.com>

To: "WIGNELL, CLIFFORD (CLIFFORD)"
<cwignell at alcatel-lucent.com>

Cc: discussion at sipforum.org

Sent: Friday, August 29, 2008 2:46:59 PM

Subject: Re: [SIPForum-discussion] No Ring Back Tone Issue 



Hi there



Beware that most operators do not allow early media as a default
configuration. Especially if you have an ISDN interconnect, it's not usual
enabled from the operators side. Meaning that "you" cannot send early
media back to the operaor.

Meaning that if a call flows trough your net like this :
(A-sub->PSTN->SBC1->...internet...->SBC2->PSTN->B-sub) Then
early media(ringtones, announcements etc) from B-subs "PSTN" will not
reach A-subs PSTN if SBC1's interconnect stops early-media...



regards

Torstein







 



On Fri, Aug 29, 2008 at 12:39 AM, WIGNELL, CLIFFORD (CLIFFORD) <cwignell at alcatel-lucent.com>
wrote: 





Hello Raghul, 

  

Look at RFC3666, it describes the
SIP<->PSTN call flows, by contrast RFC3665 provides "normal"
SIP flows; good places to start. 

  

Best regards 

  



Cliff Wignell 











From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org]
On Behalf Of Raghul Prasanna

Sent: Friday, 29 August 2008 5:33
AM

To: Vivek Batra

Cc: discussion at sipforum.org  







Subject: Re: [SIPForum-discussion]
No Ring Back Tone Issue 











  


 
  
  
  HI
  Vivek, 
  
  
    
  
  
    
  
  
   From
  your explanation I can understand that 183 with SDP result in early media,
  but incase of 180 with or without SDP with no 183, will reult in UA
  playing local ringback tone.. 
  
  
    
  
  
  Just
  want to know is there any RFC that says the above point... 
  
  
    
  
  
  Thanks, 
  
  
  Raghul

  

  --- On Wed, 27/8/08, Vivek Batra <vivek7683 at gmail.com>
  wrote: 
  
  
  From: Vivek Batra <vivek7683 at gmail.com>

  Subject: Re: [SIPForum-discussion] No Ring Back Tone Issue

  To: "ramon nolasco" <rpnolasco at yahoo.com>

  Cc: discussion at sipforum.org

  Date: Wednesday, 27 August, 2008, 4:09 PM 
  
  
  Comments inline in RED.

  

  --VB 
  
  On Wed,
  Aug 27, 2008 at 9:23 AM, ramon nolasco <rpnolasco at yahoo.com> wrote: 
  
  
  
  Hi All, 
  
  
    
  
  
  Greetings
  and a good day to all of you! I have this and quite a rare one
  to me of "no ring back tone" problem with one of
  our interconnecting partners. Partner claims that our system
  normally sends "183 Session Progress" and "180 Ringing".
  That we are sending "183 Session Progress" without the
  ringtone data though our "180 Ringing" has a ringtone data.
  That their system looks for the ringtone data to process from the first
  received message response, which is our "183 Session Progress" and
  afterward disregards our "180 Ringing" response that followed,
  wherein the ringtone data is indeed present,  thus resulting to a
  successfull call but without ringback tone.  
  
  
    
  
  
  My
  questions are, per standard: 
  
  
   When and why does "183 Session
       Progress" is being sent as a response? 
  
  
  
  
   '183 Session Progress' or referred as Early Media is
  generally sent when media (RTP) is required within early dialog. 

  Media is required in the early dialog
  when the call is placed from IP to PSTN. 

  When ITSP/ Gateway routes the call from IP to PSTN, it generally sends the
  183 response with SDP body and all the tones/ message are played by gateway
  to UA. However in case of 180 Ringing, RBT is played by local UA itself. 
  
  
  
  
  
     
   Should "183 Session Progress"
       sometimes also can replace
       "180 Ringing", thus have the ringtone data? or 
  
  
  
  
  
  Yes. It depends on the local policy of UA whether it wants to stop
  the media created with 183 Session Progress and start playing local RBT or
  discards the 180 Ringing (recieved after 183 Session Progress) and remains
  connect the media till the final response. 

  You will found lot of UA in the marked having both type of implementations.
   
  
  
    
  
  
  
  
  
     
   or it's always "180 Ringing" that has
       the ringtone data? 
  
  
  
  
  
  I am not sure what you are referring as Ringtone data.
  I believe that you are referring
  whether 180 Ringing has SDP or not.

  If you are referring the same, my answer would be Yes. 180 Ringing can have
  SDP body but this is not used to connect early media. 180 Ringing with SDP
  refers the Offer-Answer model as per RFC 3262.

  Only 183 Session Progress is sent as response to connect early media. 
  
  
  
  
  
     
   Our setup is SBC-to-SBC, Huawei Eudemon
       2300-to-Mediaring Voizbridge, is it us who really has the problem or who
       needs then to adjust? Adjust what? 
  
  
  
  
  
  In the above statements, you are referring that 183 Session
  Progress has no ringtone data. What actually you are referrig? You want to
  say that 183 Session Progress has no SDP? 

  Can you provide us the complete call
  flow? 
  
  
  
  
  
     
   Is my partner's claim of "the should be
       system flow and process" standard? 
  
  
  Appreciate
  any of your solution advice and many thanks in advance :) 
  
  
    
  
  
  Best
  regards, 
  
  
    
  
  
  Mon 
  
  
    
  
  

  _______________________________________________

  This is the SIP Forum discussion mailing list

  TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion

  Post to the list at discussion at sipforum.org 
  
  
    
  
  
  _______________________________________________This is the SIP Forum discussion mailing listTO UNSUBSCRIBE, or edit your delivery options, please visithttp://sipforum.org/mailman/listinfo/discussionPost to the list at discussion at sipforum.org 
  
 




Send instant messages to your online friends http://uk.messenger.yahoo.com
 











_______________________________________________

This is the SIP Forum discussion mailing list

TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion

Post to the list at discussion at sipforum.org 



   







   







   



 

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org



      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20080904/cd5e6589/attachment-0002.html>


More information about the discussion mailing list