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Due to the multiple access technique (FDMA), the RF signal can be received in a wideband (usually 36 MHz for a 70 MHz IF, 72 MHz for a 140 MHz IF). The relevant carrier occupies a reduced part of this bandwidth. The automatic gain control (AGC) must compensate the differences both of the spectral density (coming from the uplink) and of the global level (due to the downlink). The variable rate of carried data involves a variation of spectrum bandwidth which increases the complexity of the gain control somewhat. An AGC is based on a bandpass filter, a variable gain amplifier, a detector and a feedback loop. In modern receivers, variation of the bandwidth of the filters should be implemented by digital filtering, taking into account the constraints on linearity, the noise factor of the complete receiver and the limitation in the resolution of A/D converters. Typically, the control of gain is achieved in two steps. The first is done in an analogue way and controls the power inside a bandwidth determined by the tightest filter before A/D conversion. The second operates after the reduction of band by the digital filter.

Call hold This feature has many forms in the PSTN, from a button on a telephone set that simply cuts the speaker and microphone to advanced features in PBX or ISDN systems In a SIP network, a call is placed on hold by sending a re-INVITE, changing the media stream from bidirectional (sendrecv) to unidirectional (sendonly) Note that older SIP implementations may implement hold with a re-INVITE with a connection IP address of 0000 in the SDP The call is taken off hold when either party sends a re-INVITE with bidirectional media or a nonzero connection IP address Call park and pickup In this feature, a call is placed on hold at one location and then retrieved (picked up) at another location There are a number of ways to implement these features in SIP.

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pd = -0.5*deltat*((vol*vol)/(dx*dx) - drift/dx); // calculate coef cients for (j = -M; j <= M; j++) { S[N][j] = price*exp(j*dx); S[0][j] = price; } // compute stock prices for (i = 1; i < N; i++) { for (j = -M; j <= M; j++) S[i][j] = S[i-1][j]*exp(j*dx); } // calculate payoffs if (type == P ) { for (j = -M; j <= M; j++) { P[N][j] = max(strike - S[N][j],0); } // calculate boundary conditions for (i = 0; i < N; i++) { P[i][-M] = P[i][-M+1] + 1*(S[i][-M+1]-S[i][-M]); // derivative boundary // condition P[i][M] = 0; } } else // if type == C { // calculate boundary conditions for (j = -M; j <= M; j++) { P[N][j] = max(S[N][j] - strike,0); } // calculate boundary conditions for (i = 0; i < N; i++) { P[i][-M] = 0; P[i][M] = P[i][M-1] + (S[i][M] - S[i][M-1]); // derivative boundary condition } } for (j = -M+1; j < M; j++) b[j] = (1-a)*P[N][j] + a*(P[N][j+1] + P[N][j-1]); b[-M+1]= b[-M+1] + a*P[N][-M]; b[M-1] = b[M-1] + a*P[N][M];

Some of them use third-party call control and a re-INVITE, while others use a REFER and then a redirect Call forwarding There are three options with this feature: forward on busy, don t answer, and unconditional Forwarding can be done in SIP either in a proxy or in a user agent, as shown in 7, SIP Service Creation A proxy can translate one URI for another, resulting in a forward that is transparent to the calling user agent Alternatively, a user agent or a proxy can issue a redirection response (302 Moved) A proxy receiving a 486 Busy Here response can invoke a call-forward-on-busy feature by generating an ACK and then forwarding the INVITE to another URI.

A proxy can also start a ring timer upon receipt of a 180 Ringing response, and then send a CANCEL and proxy the INVITE to another URI to implement a call-forward, don t-answer service Calling line identification The ability to display calling line identification is a useful feature in the PSTN to aid the caller during alerting in deciding whether to answer a call or to implement automated screening services For example, a feature could be implemented to block incoming SIP calls in which the calling party has not been identified The basic functionality is built into SIP to accomplish this, using the From header However, since the From header is populated by the calling user agent and not by a trusted source such as a carrier, this calling line identification is not verified or guaranteed to be accurate.

The use of the Identity header field to provide a way to check the validity of the From header field is discussed in 9, SIP Security..

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For example, the following radially symmetric kernels satisfy the foregoing conditions: Uniform kernel: cD, KU ( x ) = 0, Epanechnikov kernel: 1 c 1 D + 2 1 x ) D( KE ( x ) = 2 0, Gaussian (normal) kernel: KN ( x ) = ( 2 )

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