[SIPForum-discussion] SIP-I and SIP-T

[vijay kant gupta]

http://www.tekelec.com/tekelec-blog/index.php/2009/05/faq-what-are-sip-i-and-sip-t/#.UUBAFDeDNtg




On Wed, Mar 13, 2013 at 11:45 AM, sri nivas <i.v.srinivas at gmail.com> wrote:
> can you share this article link, i am excited to know the detailed call flow
> from a Sip UA to a PSTN End point.
>
> Thank you.
>
> On Fri, Mar 8, 2013 at 12:42 PM, vijay kant gupta
> <vijaykant.it2002 at gmail.com> wrote:
>>
>> Hi ,
>>
>> I just Paste an article.
>>
>> Regards
>> Vijay
>>
>>
>> SIP-I and SIP-T refer to two very similar approaches for interworking
>> ISUP networks with SIP networks. In particular, they provide the means
>> for conveying ISUP-specific parameters through a SIP network so that
>> calls that originate and terminate on the ISUP network can transit a
>> SIP network with no loss of information.
>>
>> SIP-T was developed by the IETF — the same body that developed the SIP
>> protocol itself — around the same time the most recent version of SIP
>> was being developed (mid-2002). It is defined by RFC 3372, RFC 3398,
>> RFC 3578, and RFC 3204.
>>
>> SIP-I was developed by the ITU in 2004, and made use of most of the
>> constructs defined in the IETF SIP-T effort. It is defined by ITU-T
>> Q.1912.5.
>>
>> SIP-I and SIP-T both define the mapping of messages, parameters, and
>> error codes between SIP and ISUP. Both of them are fully interoperable
>> with compliant SIP network components on the SIP network.
>>
>> The key differences between SIP-I and SIP-T are:
>>
>>     SIP-I defines a mapping from SIP to BICC (in additional to ISUP),
>> while SIP-T addresses only the ISUP case, and
>>     SIP-T is inherently designed for interoperation with native SIP
>> terminals, while SIP-I is restricted for use between PSTN gateways
>> only.
>>
>> SIP-I and SIP-T also define somewhat different mappings of information
>> between the protocols, mostly in terms of converting from SIP error
>> codes to ISUP cause codes.
>>
>> The way SIP-I and SIP-T allow transparent transit of ISUP parameters
>> through a SIP network is by attaching a literal copy of the original
>> ISUP message to the SIP message at the ingress PSTN gateway; this ISUP
>> message appears as another body on the SIP message (typically, a peer
>> to an SDP body).
>>
>> The SIP network ignores the extra ISUP body, processing the SIP
>> message as it normally would. After the SIP service network performs
>> any necessary modifications to the SIP message, it arrives at the PSTN
>> egress gateway. This egress gateway uses the attached ISUP message as
>> the basis for the ISUP message it will send; however, it first makes
>> modifications necessary to match changes made to the SIP message
>> during its traversal of the SIP network.
>>
>>
>>
>> As mentioned before, with SIP-T, the messages may also terminate on
>> the native SIP terminals in the network, which will ignore the extra
>> ISUP body. Additionally, messages may originate from these SIP phones
>> and terminate on the PSTN gateways, which will then generate a new
>> ISUP message for the PSTN.
>>
>> Putting this together in a call flow, a typical successful call setup
>> from a PSTN terminal to another PSTN terminal through a SIP network
>> can look something like this:
>>
>> _______________________________________________
>> 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
>
>