[SIPForum-discussion] SIP to PSTN

[Stephen James]

Remember, that ISUP is a protocol where the far end is supposed to provide 
ringback. A SIP -> PSTN gateway should open a backwards path when sending the 
IAM and send SDP backwards when sending the 18x. We have seen problems where a 
ACM (with subscriber status of free - indicating terminating party is ringing) 
results in a 180 with SDP and the receiver of that 180 not cutting through the 
far end ringback. The ACM with subscriber free may have to be sent as a 183 
w/SDP to ensure correct interworking.
 
Stephen James 
sjames_1958 at yahoo.com
 
We are not princes of the earth, we are the descendants of worms, and any 
nobility must be earned.





________________________________
From: Abhisek Acharya <abhisek.acharya at gmail.com>
To: Suryakanta Behera <skbehera321 at gmail.com>
Cc: discussion at sipforum.org
Sent: Sun, January 20, 2013 11:23:34 PM
Subject: Re: [SIPForum-discussion] SIP to PSTN



Question--:In SIP to PSTN Call Flow In-between GW are there, Someone tell me 
after sending ACM message "from SSP to GW" then GW what  it will send 180/183, 
if it is sending 183-Res to SIP then Two-way RTP will flowing in between SIP to 
GW  and one-way media will flowing SSP to GW"

ANSWER--

1.The above one is a very very tricky situation.When ACM message comes from ISUP 
side ideally it means the phone is ringing on the ISUP side and the calling 
party should hear the ring-back tone.

2.Again the ring back can be  local ring back or remote back.if 18x response 
contains sdp body then this should be remote ring back and if it does not 
contain sdp body then this will be local ring back.

3.Also at times ACM message comes bit early than expected just to satisfy T9 
timer which is otherwise called as early ACM.In this case the called party phone 
may not be ringing.It may be in roaming (in case of called subscriber is out of 
its HLR/MSC).In that case 183 response goes to SIP side and a subsequesnt CPG 
from ISUP side will comes from ISUP side to denote ringing.


Ideally in case of 183 response two way RTP should not flow from the SIP end and 
GW is this is only for ringback.One way audio should flow.Are you sure two way 
audio is flowing from SIP entity to GW?I believe while ringback is there calling 
party is only hearing the ringback so  RTP should flow only one way.

First Question:-

"In between two user agent a C5 switch is present and one is sip user another is 
isup user , So who is responsible to convert SIP message to ISUP message in C5 
switch. Actually what C5 are doing there & What are the functionality/Element 
contain C5 switch in particular situation".

ANSWER

C stands for CLASS which means CustomLocalArea Signaling System and this is an 
ancient term since the old days of telecom when Graham bell started the legacy 
and Catagorisation started by AT&T folks where C5 stands for end switches and C4 
switches are interconnecting switches where they connect multiple C5 switches.

But in modern day telephony where various protocols are involved and 
interworking is a MUST then there is entity which is called as TANDEM switch/NGN 
switch which takes care of the SIGNALING conversion from one protocol to other 
protocol.Like ISUP to SIP.SIGNALING conversion is doen by the NGN switch and the 
media is also converted from CIRCUIT SWITCHED MEDIA to PACKETISED MEDIA and this 
is done by with the help of  MEDIAGATEWAYS.

Ideally C5 elements maintain the end point phone informations like its 
number,its balance,etc.Ideally modern day MSCs are clasic examples of CLASS5 
switches and old day telephone exchanges are also the exmaples.Please let me 
know if something is not clear here.

SSP---------NGN---------SIP

Abhisek Acharya
GS-LAB,PUNE



On Tue, Jan 15, 2013 at 5:07 PM, Suryakanta Behera <skbehera321 at gmail.com> 
wrote:

Hi,
>
>I have a question. Can Anyone help 
>
>"In between two user agent a C5 switch is present and one is sip user another is 
>isup user , So who is responsible to convert SIP message to ISUP message in C5 
>switch. Actually what C5 are doing there & What are the functionality/Element 
>contain C5 switch in particular situation".
>
>Another Question
>"In SIP to PSTN Call Flow In-between GW are there, Someone tell me after sending 
>ACM message "from SSP to GW" then GW what  it will send 180/183, if it is 
>sending 183-Res to SIP then Two-way RTP will flowing in between SIP to GW  and 
>one-way media will flowing SSP to GW"
>
>Is it Correct ! If Correct, how two-way RTP will flowing in between SIP to GW  
>and one-way media will flowing SSP to GW.
>
>
>
>Please, describe briefly about this....
>
>Thanks & Regards
>Surya
>
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