Internet Communications Using SIP : Delivering VoIP and Multimedia Services With Session Initiation Protocol {2Nd Ed.}

(Steven Felgate) #1

often provided in band in the media path (such as ring tone, busy signal,
reorder tone, and so on). These indicators are carried in a one-way speech path
that is established as soon as the called party is alerted but prior to the call
being answered. The caller hears the ring tone or busy signal and knows how
the call is progressing.
In SIP, the media path is not established until the called party answers ( 200
OK), and all call progress indicators are assumed to be carried in the SIP
responses, not in any media path (180 Ringing, 181 Call is Being For-
warded, 486 Busy Here, 503 Service Unavailable,etc.). This is not a
problem in a call from the PSTN to SIP—the gateway simply takes the SIP
response code and generates any tones or signals in the PSTN media path.
However, for SIP-to-PSTN calling, the SIP phone’s local ring-back tone gener-
ated by the receipt of a 180 Ringingresponse from the gateway masks the in-
band progress indicators being received by the gateway. The result is that the
call may fail, and the SIP caller will never hear any indication, just the locally
generated ring-back tone.
The solution is for a PSTN-to-SIP gateway to use early media and a 183
Session Progressresponse, which is used to indicate that the call is pro-
gressing, but that the user agent server (PSTN gateway) is not able to deter-
mine from signaling what is occurring, but that information may be available
in the media path. The gateway then sends the call progress tones or signals it
is receiving in the one-way speech path in the TDM channel as RTP packets to
the SIP phone. The SIP phone receiving a 183 response knows then to play
those RTP packets instead of generating local alerting, as shown in Figure 11.2.
In general, a SIP UA that receives early media (RTP that arrives before the 200
OKanswer) should stop playing any locally generated tones and play the early
media instead.
This approach works but has an unfortunate side effect in the case of a SIP
call that may have been forked to two different locations in the PSTN. The
result of this is that two 183 responses will be received, and the SIP user agent
client will have to decide which media stream to play, or whether to mix the
two together.
Note that the gateway only sends a 183 Session Progressresponse if it
is unable to determine whether ringing is occurring. For example, if the PSTN
connection is exclusively ISDN, then the alerting message can be mapped to a
180 Ringing. However, in many cases (especially where some type of non-
SS7 signaling path is present in the PSTN), the gateway will not be able to
make this determination and will send a 183 Session Progress.
A SIP-to-PSTN call flow is shown in Figure 11.3. In this case, the called SIP
user agent sends a 180 Ringingresponse to the gateway, which then seizes a
trunk in the PSTN and sends an address complete message (ACM) to the
PSTN. If the PSTN requires in-band alerting such as a ring tone, the gateway


SIP Telephony 189
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