https://support.aa.net.uk/index.php?title=VoIP_NAT&feed=atom&action=historyVoIP NAT - Revision history2024-03-29T06:14:53ZRevision history for this page on the wikiMediaWiki 1.39.5https://support.aa.net.uk/index.php?title=VoIP_NAT&diff=17179&oldid=prevAA-Andrew: /* Tips for VoIP through NAT */2024-01-30T08:45:58Z<p><span dir="auto"><span class="autocomment">Tips for VoIP through NAT</span></span></p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 08:45, 30 January 2024</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>==Tips for VoIP through NAT==</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>==Tips<ins style="font-weight: bold; text-decoration: none;"> and recommendations</ins> for VoIP through NAT==</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling to get calls working:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling to get calls working:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*'''Have your broadband with us.''' - we provide public IPs on all our connections.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable UPnP''' on routers</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable UPnP''' on routers</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable SIP ALG''' on the router (or try enabling) See [[Disable SIP ALG]] - ALG can do funky things!</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable SIP ALG''' on the router (or try enabling) See [[Disable SIP ALG]] - ALG can do funky things!</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Enable Stun''' settings on the VoIP phone, or disable stun it - stun.aa.net.uk can be used as the server name</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Enable Stun''' settings on the VoIP phone, or disable stun it - stun.aa.net.uk can be used as the server name</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]] - usually not needed for NAT, but needed if you have public IPs</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]] - usually not needed for NAT, but needed if you have public IPs</div></td>
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<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>* Avoid ISPs that do CGNAT - that is NAT within their own network. - Having multiple levels of NAT between your handset and our service can cause problems in practice. Ask your ISP to provided you with a public IP address and no CGNAT.</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>* <ins style="font-weight: bold; text-decoration: none;">'''</ins>Avoid ISPs that do CGNAT<ins style="font-weight: bold; text-decoration: none;">'''</ins> - that is NAT within their own network. - Having multiple levels of NAT between your handset and our service can cause problems in practice. Ask your ISP to provided you with a public IP address and no CGNAT.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you have two similar VoIP phones behind a router using NAT, you may have to change the port settings on one phone.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you have two similar VoIP phones behind a router using NAT, you may have to change the port settings on one phone.</div></td>
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</table>AA-Andrewhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=17178&oldid=prevAA-Andrew: /* Tips for VoIP through NAT */2024-01-30T08:44:13Z<p><span dir="auto"><span class="autocomment">Tips for VoIP through NAT</span></span></p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 08:44, 30 January 2024</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Enable Stun''' settings on the VoIP phone, or disable stun it - stun.aa.net.uk can be used as the server name</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Enable Stun''' settings on the VoIP phone, or disable stun it - stun.aa.net.uk can be used as the server name</div></td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]] - usually not needed for NAT, but needed if you have public IPs</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]] - usually not needed for NAT, but needed if you have public IPs</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>* Avoid ISPs that do CGNAT - that is NAT within their own network. - Having multiple levels of NAT between your handset and our service can cause problems in practice. Ask your ISP to provided you with a public IP address and no CGNAT.</div></td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you have two similar VoIP phones behind a router using NAT, you may have to change the port settings on one phone.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you have two similar VoIP phones behind a router using NAT, you may have to change the port settings on one phone.</div></td>
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</table>AA-Andrewhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=16231&oldid=prevAA-Andrew: /* NAT Tips */2023-02-11T09:40:23Z<p><span dir="auto"><span class="autocomment">NAT Tips</span></span></p>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>==<del style="font-weight: bold; text-decoration: none;">NAT </del>Tips==</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>==Tips<ins style="font-weight: bold; text-decoration: none;"> for VoIP through NAT</ins>==</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling:</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling<ins style="font-weight: bold; text-decoration: none;"> to get calls working</ins>:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable UPnP''' on routers</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*'''Disable UPnP''' on routers</div></td>
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</table>AA-Andrewhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=16230&oldid=prevAA-Andrew: /* NAT Tips */2023-02-11T09:39:47Z<p><span dir="auto"><span class="autocomment">NAT Tips</span></span></p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 09:39, 11 February 2023</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>If you do want to use NAT, then here are some tips if you are struggling:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*Disable UPnP on routers</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*<ins style="font-weight: bold; text-decoration: none;">'''</ins>Disable UPnP<ins style="font-weight: bold; text-decoration: none;">'''</ins> on routers</div></td>
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<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*Disable SIP ALG on the router (or try enabling) See [[Disable SIP ALG]]</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*<ins style="font-weight: bold; text-decoration: none;">'''</ins>Disable SIP ALG<ins style="font-weight: bold; text-decoration: none;">'''</ins> on the router (or try enabling) See [[Disable SIP ALG]]<ins style="font-weight: bold; text-decoration: none;"> - ALG can do funky things!</ins></div></td>
</tr>
<tr>
<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*Reduce the registration 'expiry' - try 10 minutes (600 seconds) the default on devices us usually an hour.</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Reduce the <ins style="font-weight: bold; text-decoration: none;">'''</ins>registration 'expiry<ins style="font-weight: bold; text-decoration: none;">'''</ins>' - try <ins style="font-weight: bold; text-decoration: none;">'''</ins>10 minutes<ins style="font-weight: bold; text-decoration: none;">'''</ins> (600 seconds) the default on devices us usually an hour.<ins style="font-weight: bold; text-decoration: none;"> - helps keeps the registration active</ins></div></td>
</tr>
<tr>
<td colspan="2" class="diff-empty diff-side-deleted"></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Set '''SIP Keepalive''' (if the phone has this option) to '''30 seconds''' - this helps keep the NAT session live on the router</div></td>
</tr>
<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*If the VoIP phone has a config setting to enter your external/public IP address, then enter the address of your router's WAN port</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*If the VoIP phone has a config setting to enter your external/public IP address, then enter the address of your router's WAN port</div></td>
</tr>
<tr>
<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*<del style="font-weight: bold; text-decoration: none;">Disable</del> Stun settings on the VoIP phone, or <del style="font-weight: bold; text-decoration: none;">try</del> <del style="font-weight: bold; text-decoration: none;">enabling</del> it - stun.aa.net.uk can be used as the server name</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*<ins style="font-weight: bold; text-decoration: none;">'''Enable</ins> Stun<ins style="font-weight: bold; text-decoration: none;">'''</ins> settings on the VoIP phone, or <ins style="font-weight: bold; text-decoration: none;">disable</ins> <ins style="font-weight: bold; text-decoration: none;">stun</ins> it - stun.aa.net.uk can be used as the server name</div></td>
</tr>
<tr>
<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]]</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]]<ins style="font-weight: bold; text-decoration: none;"> - usually not needed for NAT, but needed if you have public IPs</ins></div></td>
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<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
</tr>
<tr>
<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>If you have <del style="font-weight: bold; text-decoration: none;">2</del> similar VoIP phones behind a router using NAT, you <del style="font-weight: bold; text-decoration: none;">will</del> have to change the port settings on one phone.</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>If you have <ins style="font-weight: bold; text-decoration: none;">two</ins> similar VoIP phones behind a router using NAT, you <ins style="font-weight: bold; text-decoration: none;">may</ins> have to change the port settings on one phone.</div></td>
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<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>The local SIP port (often 5060) and the local RTP port(s) can't be the same on the 2 phones - if they are the same, you'll</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>The local SIP port (often 5060) and the local RTP port(s) can't be the same on the 2 phones - if they are the same, you'll</div></td>
</tr>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>get weirdness on incoming calls.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>get weirdness on incoming calls.</div></td>
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</table>AA-Andrewhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=16228&oldid=prevAA-Andrew: /* NAT Tips */2023-02-09T21:50:40Z<p><span dir="auto"><span class="autocomment">NAT Tips</span></span></p>
<table style="background-color: #fff; color: #202122;" data-mw="interface">
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">← Older revision</td>
<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 21:50, 9 February 2023</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable UPnP on routers</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable UPnP on routers</div></td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable SIP ALG on the router (or try enabling) See [[Disable SIP ALG]]</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable SIP ALG on the router (or try enabling) See [[Disable SIP ALG]]</div></td>
</tr>
<tr>
<td colspan="2" class="diff-empty diff-side-deleted"></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Reduce the registration 'expiry' - try 10 minutes (600 seconds) the default on devices us usually an hour.</div></td>
</tr>
<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*If the VoIP phone has a config setting to enter your external/public IP address, then enter the address of your router's WAN port</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*If the VoIP phone has a config setting to enter your external/public IP address, then enter the address of your router's WAN port</div></td>
</tr>
<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable Stun settings on the VoIP phone, or try enabling it - stun.aa.net.uk can be used as the server name</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Disable Stun settings on the VoIP phone, or try enabling it - stun.aa.net.uk can be used as the server name</div></td>
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</table>AA-Andrewhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=14869&oldid=prevAdsb: Mention configuring the public address, or using a STUN server to obtain it2021-01-21T14:55:33Z<p>Mention configuring the public address, or using a STUN server to obtain it</p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">← Older revision</td>
<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 14:55, 21 January 2021</td>
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<td colspan="2" class="diff-lineno">Line 7:</td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*The Technicolor broadband routers we used to supply with Home::1 provide a full SIP/NAT ALG which means they work such that neither the phone nor our call server know NAT is in use. This appears to be a well implemented ALG and just works.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*The Technicolor broadband routers we used to supply with Home::1 provide a full SIP/NAT ALG which means they work such that neither the phone nor our call server know NAT is in use. This appears to be a well implemented ALG and just works.</div></td>
</tr>
<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Using a FireBrick FB2700 providing NAT has no ALG and does simple dynamic port forwarding of outgoing UDP connections with a default timeout of over 2 minutes. This also just works as the call server recognises the NAT and sends one minute keep-alive packets to hold the NAT session open, as well as sending symmetric RTP response packets.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Using a FireBrick FB2700 providing NAT has no ALG and does simple dynamic port forwarding of outgoing UDP connections with a default timeout of over 2 minutes. This also just works as the call server recognises the NAT and sends one minute keep-alive packets to hold the NAT session open, as well as sending symmetric RTP response packets.</div></td>
</tr>
<tr>
<td colspan="2" class="diff-empty diff-side-deleted"></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Some phones allow the public IP address to be entered in the configuration (e.g. Grandstream products have a "Use NAT IP" option).</div></td>
</tr>
<tr>
<td colspan="2" class="diff-empty diff-side-deleted"></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*A STUN server enables a phone to be told its public IP address (e.g. ''stun.aa.net.uk).</div></td>
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<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Other cases where testing has been done have usually required one or other approach, and in some cases required "NAT assist" to be disabled on phones or routers to allow the correct operation.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Other cases where testing has been done have usually required one or other approach, and in some cases required "NAT assist" to be disabled on phones or routers to allow the correct operation.</div></td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings</div></td>
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</table>Adsbhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=14719&oldid=prevAdsb: Fixing the multiple phones behind NAT problem2020-11-14T15:33:04Z<p>Fixing the multiple phones behind NAT problem</p>
<table style="background-color: #fff; color: #202122;" data-mw="interface">
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 15:33, 14 November 2020</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Working perfectly some of the time, and not others</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Working perfectly some of the time, and not others</div></td>
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<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Stopping working one day with no apparent reason</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>*Stopping working one day with no apparent reason</div></td>
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<td class="diff-marker" data-marker="−"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>*Working when there is only one phone behind NAT but not when more than one</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>*Working when there is only one phone behind NAT but not when more than one<ins style="font-weight: bold; text-decoration: none;">. This is caused by muliple phones using the same local SIP port, and/or the same local RTP port. Each phone should use a different local SIP port, and a different local RTP port or non-overlapping local RTP port ranges.</ins></div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>===What is the solution===</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>===What is the solution===</div></td>
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</table>Adsbhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=14718&oldid=prevAdsb: Tiny typo2020-11-14T15:27:02Z<p>Tiny typo</p>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>We supply various equipment. Usually it is equipment made by third parties, such as ZyXEL, Billion, etc. They will often include features such a NAT in the equipment, which you can use. NAT is not something that has a defined standard to which it works - it is a bodge, and so the quality of the NAT features in such equipment is not something we can guarantee. We supply them as is in that respect, and you are welcome to ask the manufacturers for details of what they do with NAT before you purchase (but don't expect a detailed reply). Our recommendation is, as always, not to use NAT at all. Using any sort of NAT, whatever the manufactuters claim, is likely to cause problems with some protocols, especially SIP.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>We supply various equipment. Usually it is equipment made by third parties, such as ZyXEL, Billion, etc. They will often include features such a NAT in the equipment, which you can use. NAT is not something that has a defined standard to which it works - it is a bodge, and so the quality of the NAT features in such equipment is not something we can guarantee. We supply them as is in that respect, and you are welcome to ask the manufacturers for details of what they do with NAT before you purchase (but don't expect a detailed reply). Our recommendation is, as always, not to use NAT at all. Using any sort of NAT, whatever the manufactuters claim, is likely to cause problems with some protocols, especially SIP.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>We also supply FireBrick firewall/routers. These do allow basic IP and TCP/<del style="font-weight: bold; text-decoration: none;">IDP</del> port mapping if you want, and as such can provide basic NAT features. In the case of the FireBrick there are specifically no ALGs (Application Layer Gateways) and any NAT or mapping is purely at the TCP/UDP level. You can configure timeouts for sessions in the configuration for FireBricks which can help. Again, our recommendation is, as always, not to use NAT at all. If your VoIP devices have fixed IP addresses then firewall rules on a FireBrick are simple to set up to allow correct SIP operation.</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>We also supply FireBrick firewall/routers. These do allow basic IP and TCP/<ins style="font-weight: bold; text-decoration: none;">UDP</ins> port mapping if you want, and as such can provide basic NAT features. In the case of the FireBrick there are specifically no ALGs (Application Layer Gateways) and any NAT or mapping is purely at the TCP/UDP level. You can configure timeouts for sessions in the configuration for FireBricks which can help. Again, our recommendation is, as always, not to use NAT at all. If your VoIP devices have fixed IP addresses then firewall rules on a FireBrick are simple to set up to allow correct SIP operation.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>===NAT is evil===</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>===NAT is evil===</div></td>
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</table>Adsbhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=14717&oldid=prevAdsb: Add explanation to the NAT problem2020-11-14T15:25:38Z<p>Add explanation to the NAT problem</p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 15:25, 14 November 2020</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Consider a simple access to a web site - your machine (behind NAT) makes a connection to a web server and requests a page which is sent in that connection. The connection (using TCP) can be NATted easily allowing two way data flow and allowing the web page to be received.</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Consider a simple access to a web site - your machine (behind NAT) makes a connection to a web server and requests a page which is sent in that connection. The connection (using TCP) can be NATted easily allowing two way data flow and allowing the web page to be received.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>However, telephony is different. For a start the process is broken in two - signalling and media. This means the "connection" to establish a call is different to the actual audio that makes up the call. When the <del style="font-weight: bold; text-decoration: none;">signally</del> sets up the call it says where to send the audio - except it will be saying "send it to 192.168.1.124" (i.e. a NATted address). That won't work because that is a private IP address and not reachable from the Internet.</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>However, telephony is different. For a start the process is broken in two - signalling and media. This means the "connection" to establish a call is different to the actual audio that makes up the call. When the <ins style="font-weight: bold; text-decoration: none;">signalling</ins> sets up the call it says where to send the audio<ins style="font-weight: bold; text-decoration: none;"> (see the above SIP trace)</ins> - except it will be saying<ins style="font-weight: bold; text-decoration: none;"> (for example)</ins> "send it to 192.168.1.124" (i.e. a NATted address). That won't work because that is a private IP address and not reachable from the Internet.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>There are <del style="font-weight: bold; text-decoration: none;">two</del> answers to this problem:</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>There are <ins style="font-weight: bold; text-decoration: none;">three</ins> answers to this problem:</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div># Tell the phone what its externally visible (public) IP address is. Some phones allow you to enter this in the configuration. If not, you can use a STUN server - which tells the phone what its externally visible IP address is, and attempts to tell the phone about the firewall it's behind. You can use ''stun.aa.net.uk'' as the STUN server.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div># Tell the phone what its externally visible (public) IP address is. Some phones allow you to enter this in the configuration. If not, you can use a STUN server - which tells the phone what its externally visible IP address is, and attempts to tell the phone about the firewall it's behind. You can use ''stun.aa.net.uk'' as the STUN server.</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div># Rely on the modem/router having an Application Layer Gateway (ALG) which will rewrite the packet to change the private IP address to the modem/router's externally visible IP address. And which does it properly - the challenge is that it's often very difficult to see what the ALG has done to the packet (need to be able to look at packets going down the phone line).</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div># The call server can break the SIP specification and wait for the audio to come from the device and send its audio back to the same place - i.e. allow the phone to make an outgoing connection for the audio first. The SIP standard does not work like that, and indeed there is no requirement for the phone behind NAT to expect its audio to come back to the same IP and port from which is sends it. Of course many phones do this as it happens, and they might work in such cases, but some do not and they don't work. You have no way to tell when buying a phone, or upgrading its software. It is quite valid for the phone to stop working with your NAT and still meet the standard for SIP.</div></td>
<td class="diff-marker" data-marker="+"></td>
<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div># The call server can break the SIP specification and wait for the audio to come from the device and send its audio back to the same place - i.e. allow the phone to make an outgoing connection for the audio first. The SIP standard does not work like that, and indeed there is no requirement for the phone behind NAT to expect its audio to come back to the same IP and port from which is sends it. Of course many phones do this as it happens, and they might work in such cases, but some do not and they don't work. You have no way to tell when buying a phone, or upgrading its software. It is quite valid for the phone to stop working with your NAT and still meet the standard for SIP<ins style="font-weight: bold; text-decoration: none;">. Of course, as a user you have no control over how the call server works</ins>.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>The other big problem is telephone calls are both ways - you don't just make calls but you receive them. This means the phone registers where it is. I.e. it sends a message saying "I can be reached on 192.168.1.124", which won't work. Now, again, some call servers will break the SIP standard, ignore the actual stated connection details and assume calls can be sent to the IP and port from which the registration came. This may work as long as the session that did the registration is still active on the NAT device. The device will have a timeout. It could be seconds, minutes, hours - no way to tell on most NAT devices. So that means incoming calls work some of the time (within X seconds of the last periodic registration message). Now, some call servers will send a dummy message every few seconds to keep the session active. Some phones may do that. They are both guessing what the NAT device does with timeouts, and may guess right. This is, of course, non standard for phone and call server to do. It's all very well for the phone to be listening on, for example, port 5060 for control messages ("incoming call") and ports 5008 - 5012 for audio - but if the firewall doesn't know about these listening ports the call isn't going to succeed. The best solution is to have firewall rules which allow VoIP packets from the AAISP VoIP servers - see [[VoIP Firewall]]</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>The other big problem is telephone calls are both ways - you don't just make calls but you receive them. This means the phone registers where it is. I.e. it sends a message saying "I can be reached on 192.168.1.124", which won't work. Now, again, some call servers will break the SIP standard, ignore the actual stated connection details and assume calls can be sent to the IP and port from which the registration came. This may work as long as the session that did the registration is still active on the NAT device. The device will have a timeout. It could be seconds, minutes, hours - no way to tell on most NAT devices. So that means incoming calls work some of the time (within X seconds of the last periodic registration message). Now, some call servers will send a dummy message every few seconds to keep the session active. Some phones may do that. They are both guessing what the NAT device does with timeouts, and may guess right. This is, of course, non standard for phone and call server to do. It's all very well for the phone to be listening on, for example, port 5060 for control messages ("incoming call") and ports 5008 - 5012 for audio - but if the firewall doesn't know about these listening ports the call isn't going to succeed. The best solution is to have firewall rules which allow VoIP packets from the AAISP VoIP servers - see [[VoIP Firewall]]</div></td>
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</table>Adsbhttps://support.aa.net.uk/index.php?title=VoIP_NAT&diff=14716&oldid=prevAdsb: Be more explicit about the local SIP port2020-11-14T15:15:55Z<p>Be more explicit about the local SIP port</p>
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<td colspan="2" style="background-color: #fff; color: #202122; text-align: center;">Revision as of 15:15, 14 November 2020</td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>VoIP using SIP has a control channel, which is used for registering with the VoIP server and for initiating outgoing calls and notification of incoming calls. The VoIP server normally listens on UDP port 5060. When a call is being setup and when the call is active, an additional audio channel is used. </div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>VoIP using SIP has a control channel, which is used for registering with the VoIP server and for initiating outgoing calls and notification of incoming calls. The VoIP server normally listens on UDP port 5060. When a call is being setup and when the call is active, an additional audio channel is used. </div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>When your VoIP phone places a call, a message is sent on a control channel - this will conventionally be to the VoIP server's IP address on port 5060. When the call is being setup, the two ends of the link have a conversation to agree on how to do the audio link (and video link too, if supported). The two ends of the link will agree on audio codecs to use (how the audio will be transmitted), and each end tells the other end where to send the audio packets to. The audio travels on a different channel from the control channel (using Real Time Protocol - RTP), and your VoIP phone will send something like "send your audio to this IP address, I'm listening for RTP on (for example) <del style="font-weight: bold; text-decoration: none;">ports 5008 -</del> 5012". Each end agrees, and you have a successful phone call.</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>When your VoIP phone places a call, a message is sent on a control channel - this will conventionally be to the VoIP server's IP address on port 5060<ins style="font-weight: bold; text-decoration: none;">. Your VoIP phone will use a fixed local SIP port number - this is remembered by the VoIP server because it's where incoming call messages will be sent to</ins>. When the call is being setup, the two ends of the link have a conversation to agree on how to do the audio link (and video link too, if supported). The two ends of the link will agree on audio codecs to use (how the audio will be transmitted), and each end tells the other end where to send the audio packets to. The audio travels on a different channel from the control channel (using Real Time Protocol - RTP), and your VoIP phone will send something like "send your audio to this IP address, I'm listening for RTP on (for example) <ins style="font-weight: bold; text-decoration: none;">port</ins> 5012". Each end agrees, and you have a successful phone call.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>An incoming call works the same way. It's important to note that the VoIP server will send control messages to your local SIP port,</div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>An incoming call works the same way. It's important to note that the VoIP server will send control messages to your local SIP port,<ins style="font-weight: bold; text-decoration: none;"> and your firewall connection tracker may have timed out the connection to the control channel if you haven't re-registered recently, or made an outgoing call recently - you either need to provide firewall rules to accept this control channel traffic, or send SIP Keep-Alives every (say) 120 seconds.</ins></div></td>
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<td style="color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>and your firewall connection tracker may have timed out the control channel if you haven't re-registered recently, or made an outgoing call recently - you either need to provide firewall rules to accept this control channel traffic, or send SIP Keep-Alives every (say) 120 seconds.</div></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
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<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><br /></td>
</tr>
<tr>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Here's an (edited) SIP trace of an outbound call being setup from 02083xxxxxx to 07973xxxxxx. The phone on public IP address 81.187.xx.xx sends:</div></td>
<td class="diff-marker"></td>
<td style="background-color: #f8f9fa; color: #202122; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Here's an (edited) SIP trace of an outbound call being setup from 02083xxxxxx to 07973xxxxxx. The phone on public IP address 81.187.xx.xx sends:</div></td>
</tr>
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</table>Adsb