Show Mobile Navigation

Wednesday, March 16, 2016

, , , ,

How a SIP phone places a telephone call to a PSTN through PSTN gateway.

Leke Oyetoke - Wednesday, March 16, 2016

How a SIP phone places a telephone call to a PSTN through PSTN gateway.

Given below is a step-by-step explanation of all the process that takes place while placing a call from a SIP phone to PSTN.

1. Initially, SIP phone collects the dialed digits and puts them into a SIP URI used in the Request-URI and the To header. The caller may have dialed either the globalized phone number 1-202-555-1313 or a local number 555-1313, and the SIP phone added the assumed country code and area code to produce and generate the globalized URI using the built-in dial plan.

2. The SIP phone has been preconfigured with the IP address of the PSTN gateway, so it is able to send the INVITE immediately and directly to gw.carrier.com.

3. The gateway starts the call into the PSTN by selecting an SS7 ISUP trunk to the next telephone switch in the PSTN.

4. The dialed digits from the INVITE are mapped into the ISUP IAM. The ISUP address complete message (ACM) is sent back by the PSTN to show that the trunk has been seized.

5. In this example, ringtone is produce and generated by the far-end telephone switch. The gateway maps the ACM to the 183 Session Progress responses havingan SDP showing the RTP port that the gateway will use to bridge the audio from the PSTN.

6. After reception of the 183, the caller’s UAC begins receiving the RTP packets sent from the gateway and presents the audio to the caller so they know that the call is progressing in the PSTN.

7. The call completes and finishes when the called party answers the telephone, which causes the telephone switch to send an answer message (ANM) to the gateway.

8. The gateway then cuts the PSTN audio connection through in both directions and sends a 200 OK response to the caller. As the RTP media path is already established, the gateway echoes the SDP in the 183 but causes no changes to the RTP connection.

9. The UAC sends an ACK to complete the SIP signaling exchange. As there is no equivalent message in ISUP, the gateway absorbs the ACK.

10. The call ends when the caller sends the BYE to the gateway. The gateway maps the BYE to the ISUP release message (REL).

11. The gateway sends the 200 OK to the BYE and receives an RLC from the PSTN.


0 comments:

Post a Comment