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I’ll be honest. I didn’t know there was a difference.
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<div>I’m guessing a SIP trunk to a third party app that is reported as down due to to sip option ping really is down and not some silly networking issue where an icmp ping was failing. </div>
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<div>This is good to know. </div>
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<div>And the last thing I will learn this year. ;)</div>
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On Dec 20, 2018, at 1:01 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div>Erik,</div>
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That's an interesting insight. It kind of sounds like you think ICMP Ping and SIP OPTIONS Ping are related, but they are completely different.
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<div>Just because you cannot ICMP Ping the SIP Peer at L3, doesn't mean you cannot OPTIONs them.</div>
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<div>Am I understanding your thought process correctly?</div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 11:53 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com">ryanhuff@outlook.com</a>> wrote:<br>
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Thanks,</div>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Ryan Huff, CCDP, CCNP<br>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Cisco Certified Network and Design Professional</div>
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<div id="gmail-m_-5130095315229186341divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:46 PM<br>
<b>To:</b> Erik Anderson<br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over</font>
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Not sure what kind of code you're working with but if its modern, you could try server groups. Here is a snippet from one of mine (using AT&T admitidly), sanitized for the NSA ...</div>
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<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">voice class server-group 100</span><span><br>
</span></i>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 3</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 12.x.x.x preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description PSTN SIGNALING PEERS</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class server-group 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 10.x.x.x preference 3</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">10.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">10.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description CUCM SIGNALING PEERS</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class sip-options-keepalive 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description PSTN HEARTBEAT</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class sip-options-keepalive 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description CCM HEARTBEAT</span><br>
</i></div>
<i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">{ .. other config .. }</span></i></div>
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<i><span style="font-size:8pt;line-height:normal"><br>
</span></i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">dial-peer voice 100 voip</span><span><br>
</span></i>
<div><i><span style="font-size:8pt;line-height:normal"> description INGRESS/EGRESS WITH PSTN</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> translation-profile outgoing PLUS1_STRIP</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> huntstop</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination-pattern A</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session protocol sipv2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session server-group 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination dpg 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> incoming uri via PSTN</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class codec 1 </span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-ping 60</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip profiles 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-keepalive profile 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind control source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind media source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> dtmf-relay rtp-nte sip-notify</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> no vad</span></i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">dial-peer voice 200 voip</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description INGRESS/EGRESS WITH CUCM</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> translation-profile outgoing PLUS1_STRIP</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> huntstop</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination-pattern A</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session protocol sipv2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session server-group 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination dpg 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> incoming uri via CUCM</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class codec 1 </span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip profiles 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-keepalive profile 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind control source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind media source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> dtmf-relay rtp-nte sip-notify</span><br>
</i></div>
<i><span style="font-size:8pt;line-height:normal"> no vad</span></i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">!</span></i><br>
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Thanks,</div>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Ryan Huff, CCDP, CCNP<br>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Cisco Certified Network and Design Professional</div>
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<div id="gmail-m_-5130095315229186341x_divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Erik Anderson <<a href="mailto:erik.anderson.85@gmail.com" target="_blank">erik.anderson.85@gmail.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:37 PM<br>
<b>To:</b> Ryan Huff<br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over</font>
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<div dir="ltr">Ryan,
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<div>Level 3 does not support options ping. If i try to ping the call control IP it will always fail. There is a separate pingable address, but I didnt think i could configure the options ping to use any address other than the target. </div>
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<div class="gmail-m_-5130095315229186341x_x_gmail_quote">
<div dir="ltr">On Thu, Dec 20, 2018 at 11:34 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<br>
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Couldn't you just use voice class sip options/keepalives to mark when the ITSP is down, so CUCM marks the trunk out of service and fails to the next route group member immediately (ideally, your secondary CUBE)? Seems like thats a more natural way of doing
it versus using IP SLAs...<br>
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Thanks,</div>
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- Ryan<br>
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<font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>> on behalf of Erik Anderson <<a href="mailto:erik.anderson.85@gmail.com" target="_blank">erik.anderson.85@gmail.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:03 PM<br>
<b>To:</b> cisco-voip voyp list<br>
<b>Subject:</b> [cisco-voip] SIP Fail over</font>
<div> </div>
</div>
<div>
<div dir="ltr">Morning Folks,
<div><br clear="all">
<div>
<p style="margin:0in;font-family:Calibri;font-size:11pt">We have implemented a new SIP solution with Level 3 and found that we have outbound calling failover issues. When a CUBE loses its ability to talk to its Level 3 Peer, but can still talk to CUCM outbound
calls will still connect to the CUBE, but fail connecting to Level 3. In turn CUCM still thinks the call is connected since the CUCM SIP trunk remains up to the CUBE.</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Architecture Notes:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 Locations with 1 CUBE Each</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 CUCM SIP Trunks with each connecting to one of the 4 CUBEs</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 CUCM Route Groups with Various CUBE/SIP Trunks assigned a Distribution Algorithm of
<span style="font-style:italic">Top Down</span></p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Each CUBE has 2 SIP Peers</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Each CUBE can only talk to its respective SIP peer via its local Level 3 Transport to reduce call control latency by not allowing it to use the DMVPN backup network</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Level 3 does not support SIP Options Ping</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">CUCM Trunks have SIP Options Ping enabled</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Call Flows:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Working Flow:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes</p>
<p style="margin:0in 0in 0in 3.75in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 3.75in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">CUBE Failure:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----></p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> Route Group Member #1 ----> CUBE SIP TRUNK --X--> CUBE (CUCM Cant Reach CUBE)</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> CUCM Routes Call to Next Route Group Member</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt"> Route Group Member #2 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes</p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Level 3 Transport Failure/SIP Server Failure:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----></p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE --X--> Level 3 Transport (CUBE Cant Reach Level 3 SIP Server)</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> CUCM Thinks Call Connects since the CUBE accepts the call, Phone gets dead air, never tries the next RG Member</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">My idea to fix this is to use an IPSLA to ping the pingable address on the Level 3 SIP Servers. If both address are unreachable then shutdown the CUCM Dial-Peers. This doesn’t sounds like the best way
of fixing it, but it should work. </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">If any has any other better ideas please let me know.</p>
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-- <br>
<div dir="ltr" class="gmail-m_-5130095315229186341x_x_gmail-m_-6726971829579282007x_gmail_signature">
Erik Anderson</div>
</div>
<div class="gmail-m_-5130095315229186341x_x_gmail-m_-6726971829579282007x_gmail_signature">
Telecom Manager</div>
<div class="gmail-m_-5130095315229186341x_x_gmail-m_-6726971829579282007x_gmail_signature">
Some Random Corp.</div>
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</div>
-- <br>
<div dir="ltr" class="gmail-m_-5130095315229186341x_x_gmail_signature">Erik Anderson</div>
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