Linksys SPA2002-ER User Guide - Page 54

Symmetric RTP, 13. Configuration Examples and Call Scenarios

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affects both ringing signal and call-waiting tone. Unlike the setting under the [User 1] tab, the parameter under the [PSTN User] tab has the extra choice of "Follow Line 1", which means to follow the Line 1 ringer settings (including distinctive ringing rules) instead of forcing it to use a particular ring cadence. When the SPA inserts the decoded PSTN caller-id name and number into the SIP INVITE message sent to the VoIP1 interface, it also prepends the name and number by the prefixes configured in and respectively. If the PSTN caller hangs up before Line 1 or the VoIP gateway answers the call, the Line 1 phone may continue to ring a little longer since it takes a few seconds for the SPA to realize that the PSTN line has indeed stopped ringing. This delay can be modified by setting the value (default is 5s). The configuration parameters mentioned in this section are: - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN User] 4.12. Symmetric RTP In a normal VoIP connection, the SPA sends RTP packets to the destination as specified in the SDP sent by the VoIP peer. When is set to "yes", however, SPA will change the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the SPA. This setup can facilitate symmetric NAT traversal at the remote end. The parameter is available under both [Line 1] and [PSTN Line] tabs. - [Line 1] - [PSTN Line] 4.13. Configuration Examples and Call Scenarios In this section we walk through some typical scenarios where the SPA-3000 can be applied. 4.13.1. Setup VoIP1 and VoIP2 With Separate VoIP Accounts You have 2 FWD (Free World Dialup) accounts A and B, where A is for regular VoIP service and B dedicated for gateway functions. a) Without Using STUN or NAT Mapping [SIP] = no [SIP] = no [Line 1] = yes [Line 1] = 5060 [Line 1]

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© 2003 - 2004 Sipura Technology, Inc
Proprietary (See Copyright Notice on Page 2)
54
affects both ringing signal and call-waiting tone. Unlike the <Default Ring> setting under the [User 1]
tab, the <Default Ring> parameter under the [PSTN User] tab has the extra choice of “Follow Line 1”,
which means to follow the Line 1 ringer settings (including distinctive ringing rules) instead of forcing it
to use a particular ring cadence.
When the SPA inserts the decoded PSTN caller-id name and number into the SIP INVITE message
sent to the VoIP1 interface, it also prepends the name and number by the prefixes configured in
<PSTN CID Number Perfix> and <PSTN CID Name Prefix> respectively.
If the PSTN caller hangs up before Line 1 or the VoIP gateway answers the call, the Line 1 phone
may continue to ring a little longer since it takes a few seconds for the SPA to realize that the PSTN
line has indeed stopped ringing. This delay can be modified by setting the <PSTN Ring Timeout>
value (default is 5s).
The configuration parameters mentioned in this section are:
-
[PSTN Line]<PSTN Ring Thru Line 1>
-
[PSTN Line]<PSTN Ring Thru Delay>
-
[PSTN Line]<PSTN Ring Thru CWT Delay>
-
[PSTN Line]<PSTN Ring Timeout>
-
[PSTN Line]<PSTN Answer Delay>
-
[PSTN Line]<PSTN CID For VoIP CID>
-
[PSTN Line]<PSTN CID Name Prefix>
-
[PSTN Line]<PSTN CID Number Prefix>
-
[PSTN Line]<PSTN Ring Timeout>
-
[PSTN User]<Default Ring>
4.12. Symmetric RTP
In a normal VoIP connection, the SPA sends RTP packets to the destination as specified in the SDP
sent by the VoIP peer. When <Symmetric RTP> is set to “yes”, however, SPA will change the
destination to send RTP packets to the source IP address and port of the inbound RTP packet last
received by the SPA. This setup can facilitate symmetric NAT traversal at the remote end. The
parameter is available under both [Line 1] and [PSTN Line] tabs.
-
[Line 1] <Symmetric RTP>
-
[PSTN Line] <Symmetric RTP>
4.13. Configuration Examples and Call Scenarios
In this section we walk through some typical scenarios where the SPA-3000 can be applied.
4.13.1.
Setup VoIP1 and VoIP2 With Separate VoIP Accounts
You have 2 FWD (Free World Dialup) accounts A and B, where A is for regular VoIP service and B
dedicated for gateway functions.
a) Without Using STUN or NAT Mapping
[SIP]<STUN Enable> = no
[SIP]<Substitute VIA Addr> = no
[Line 1]<Line Enable> = yes
[Line 1]<SIP Port> = 5060
[Line 1]<NAT Mapping Enable] = no
[Line 1]<NAT Keep Alive Enable] = no