So if the packets are marked DF (Do not fragment) have you considered changing the source of the packets MTU size? IE Callmanager or CUCM Expressway gateway. You can changed this in the router configuration as well. Note changing a CUCM MTU size will cause the network interface to restart. I have not look at your pcap but what is your full topology, and why can your provider not accept default MTUs? They might be able to change on there side... if you are using default MTUs should be something they support if the traffic is making it to them and they are truely the ones dropping the packets.
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Thanks for follow up. Great +5 follow on changes my idea of the subject. I have followed Paul Giralts python scripts to pull RIS DB status via api. Fun stuff. I think where my understanding of the RIS restart came from is that even if I leave this at the default of 72 hours and the RIS dbservice is restarted that original information would be removed so the if a phone was unregistered within that time period that information was lost. I could be wrong about that mechanism but I know I have had issue where RIS db was restarted and phones that were once unregistered were listed as unknown afterwards.
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If you don't have any access list applied to an interface all you need to do is properly configure the voice session voip and cube commands to enable it. Note the trust list below would be to be the source of the traffic. What is your true topology look like that might help with a better answer. voice service voip allow-connections sip to sip ip address trusted list
ipv4 <ipv4-address> [<ipv4 network-mask>] Who is your ITSP? If any of the major players you can find sample configuration here. https://www.cisco.com/c/en/us/solutions/enterprise/interoperability-portal/networking_solutions_products_genericcontent0900aecd805bd13d.html SIP dial peers on the router are UDP by default I believe you can change them to TCP if you want.
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Chris got any doc that goes over the 24 hour period. I thought this only changed to unknown after the RISDB services was restarted itself.
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The answer is yes liked stated above. It is layer 2 mcast.... see comment above about igmp snooping. Show your exact model and licenses you ordered on the switch if you need someone to comment on it. The IP services model of licensing is a pretty old one, but I no longer quote licensing alot.
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It is layer 2 multicast at that point and uses IGMP snooping. Look at the license you have supported do you supported IGMP snooping and layer 2 mcast? Similar Thread https://community.cisco.com/t5/switching/ip-base-license/td-p/2579539
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You can find the similar guide like the one I sent if there is no mention of http then nope it doesn’t do it. Newer phones would be like 7800 and 8800 series. Not sure exactly when it was introduced.
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I think another way would be to look at show call active voice brief. This can show you both legs for a call based on the ID. So you can see starting codec on one call leg and ending codec on another. Have not tested myself but i would imagine give that a try.
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HTTP Download Enhances the file download process to the phone to use HTTP by default. If the HTTP download fails, the phone reverts to using the TFTP download.
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Newer phones actually use http instead of TFTP. TFTP config files can be encrypted during download. Also CUCM has a ITL file that is used to verify the source of where the config file comes from after the first config file is received. Since you security team is bugging you make sure you tell them that you updated your phone firmware to version 12.7 to remove the CDPWN problem. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8800-series/english/adminguide/P881_BK_C136782F_00_cisco-ip-phone-8800_series/P881_BK_C136782F_00_cisco-ip-phone-8811-8841_chapter_01011.html https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8800-series/english/adminguide/P881_BK_C136782F_00_cisco-ip-phone-8800_series/P881_BK_C136782F_00_cisco-ip-phone-8811-8841_chapter_01011.html
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There was an old bug ID / feature request to change this so that the tone plays more than 15 seconds apart. I think other third party recording solutions would take away this limitation but your going to pay more.
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If you don't enable it, the trunk will only run on devices within the devices in the CUCM group. If checked it will run on all active nodes with the cucm service enabled.
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