cisco cube rtp ports

I am not sure about the RTP range used by Avaya.The RTP port range used by Cisco is 16384 - 32767. In some versions of IOS, you can whitelist SIP IPs as follows: In global configuration mode. It should not matter. Follow Us. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 510647 510648 17882 10012 X.X.X.6 X.X.X.1 2 510648 510647 17884 12818 Y.Y.Y.68 Y.Y.Y.147 Found 2 active RTP connections Your Cisco CUBE configured with any internal setup to your Cisco Call Manager and any network connectivity you need to allow your users to dial. Symptom: sip provider--sip--CUBE--sip--CUCM8.1--sip‹rightfax(RF) Steps : 1. This SIP trunk is part in a route list for route pattern 9.01753123123 On the CUBE Router we have the following Dial Peer and respective voice translation profiles. Important note: If the other party uses MXP series TelePresence, then the ports differ, for example RTP media ports for MXP series are UDP 46000-49000 and not 2326-2485. Subject: [cisco-voip] FW: Cisco CUBE Sip to Sip Question Hi All Hopefully an easy couple of question, In Communications Manager we have created a SIP trunk to our CUBE router. First try, no luck. -Is it sufficient if I open ports TCP/UDP 5060/5061(SIP) and UDP range 16384-32767(RTP) between our CUBE and client CUCM cluster/Service provider SBC ? SRST phone registration procedure uses the translation pattern in transformation mask how phone get registered. Make sure that the port range is large enough for anticipated number of concurrently recorded calls. Most Cisco documentation specifies that RTP & RTCP traffic will use a dynamically chosen port number in the range 16384 to 32767, with RTP using an even port number & RTCP using the subsequent odd numbered port. quick question is it mandatory to open all RTP range ports from 16384 to 32766 from the firewall is there anyway to force telepresence end points to use lower range of ports than that?? As per the below document the RTP port range used by … - Client want to know what UDP port range should be allowed on there firewall to allow traffic from the CUBE. ITSP side responded the call with 183/200OK with rtp-nte. Make sure that the port range is large enough for anticipated number of concurrently recorded calls. So you need to know about the other party equipment to open the required ports in the firewall. edit: I'm not sure show IP Interface brief commands will work, The MDS9000 is a SAN fiber switch, not a normal workstation switch. Some devs seem to pick a low port all the time, some pick different. of current calls SIP-UA show sip-ua calls br (Vz IP address and number of calls) Cisco UCSC-C240-M3S VMWare host running ESXi 5.5 Standard Cisco ISR4431/K9 router as CUBE Cisco ISR4431/K9 (1RU) processor with 1684579K/6147K bytes of memory with 4 Gigabit Ethernet interfaces Cisco 2851 Fax Gateway IP phones 9971 (SIP) and 8945 (SIP) Cisco 3945 router for hardware Conference Bridge Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate! I moved my modified desktop view xml file over and restored the default. Cisco CUBE (Cisco Unified Border Element) Debugging and Show Commands. Just allow these ports on your firewall along with the standard udp range (16384 - 32767). You wouldn’t want every SIP client out there to send invites to your CUBE, using it as a proxy to call whoever he wishes. Unlike Expressway, >From all the devices. In some versions of IOS, you can whitelist SIP IPs as follows: In global configuration mode. Configure Cisco CUBE SIP Options Ping. 1 Refers to a pre-configured ordered list of codecs. This allows the VoIP RTP layer to safely drop packets without proper sessions (phantom packets) received on these ports of the Cisco Unified Border Element (CUBE) or Voice time-division multiplexing (TDM) gateways. 8000 - 48198 is the range supported by ISR-4k and also ASR routers. The Route Processor 3 adds more options for higher performance, memory, and storage to the ASR 1000 Series. If I adjust the CUBE configuration such that media (RTP) flows around the CUBE router (ie RTP flows directly between the Cisco IP Phone and the ISP SBC) I get full duplex audio. callID(18446744073709551615), port(38164) socket(0x0) Topology: PhoneA----CUCM-----(CUBE)---- … Issue is when the call lands on CUBE 1 it goes to CUCM-1 and user answers the phone. of current calls SIP-UA show sip-ua calls br (Vz IP address and number of calls) show sip-ua calls summary (number of calls) ... (IP and ports of CUBE--phone rtp stream) sh call threshold (stats | config) - Show incoming call threshold and num. Infact some of cisco's product do not use the standard udp port range eg Cisco VCS servers. show udp | i (IP and ports of CUBE–phone rtp stream)!– H323/ISDN debug voice ccapi inout debug voice dialpeer debug isdn q931 debug voip ccapi inout debug h245 asn1 (dtmf) debug voip rtp session named-event (dtmf) debug voice rtp session named-event (dtmf) RTP Port Range: Provides the capability where the port range is managed per IP address range. Will modifying the range affect other SIP connections on the CUBE? The router will just stream the RTP to that port. Control h323 = tcp/1720. This features solves the problem of limited number of rtp ports for more than 4000 calls. Device# show voip rtp connection VoIP RTP Port Usage Information: Max Ports Available: 8091, Ports Reserved: 101, Ports in Use: 2 Port range not configured, Min: 16384, Max: 32767 Ports Ports Ports Media-Address Range Available Reserved In-use Default Address-Range 8091 101 2 VoIP RTP active connections : No. It enables combination of an IP address and a port as a unique identification for each call. show cdp neighbor will show attached devices, not ports. Media= udp(rtp) / 16384 to 32767. I want to open firwall ports for traffic between our Cisco CUBE and 1.clients Cisco CallManager Cluster and 2.service provider SBC. What your VoIP provider uses for RTP does not need to be part of what IOS supports. This ACL is applied to the WAN port on the router facing the ISP. show interface status will show connected ports and their port mode. This behavior causes one-way audio as the CUBE stops sending RTP to the negotiated Media IP address and starts sending RTP to previously negotiated media IP address and port number. CUBE send EO to ITSP side . Stay connected to Research Triangle Park. Do check that these ports are open in each direction, as RTP streams are independent of each other and unidirectional. You'd have to try it on IOS. How do they negotiate RTP port numbers? cisco-rtp Cisco Proprietary RTP h245-alphanumeric DTMF Relay via H245 Alphanumeric IE h245-signal DTMF Relay via H245 Signal IE rtp-nte RTP Named Telephone Event RFC 2833 종료 종료의 요구 사항에 따라 다이얼 피어당 둘 이상의 방법을 구성할 수 있습니다. Global availability and Cloud Connected PSTN options for Cis... http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/port/8_0_2/portlist802.html. Recently upgraded to UCCX 12.5 and the longest call in queue data field is missing. As you only need 2 RTP ports per conversation (1 port per direction) I only enabled 11 ports on the router for forwarding and then used the same 11 in the ATA. Nmap port scan shows these ports as closed. UDP Port 5060-5082 range, SIP communications. of current calls SIP-UA show sip-ua calls br (Vz IP address and number of calls) Does it work? You can define your rtp port range to values you want. The ASR 1001-HX has 4 built-in 10 GE ports, 8 1 GE ports, and 4 configurable 10 GE or 1 GE ports. 3. Now, since the security guys would rarely be happy to open ~32k ports, ... (919) 392-2000 Fax: (919) 549-7201 Twitter: @CiscoSystems Mailing Address: PO Box 14987 RTP, NC 27709. do I need to open the full UDP port range, 16384 - 32767 does CM and phones use every port in this range or could I reduce it to say the first 500 , does it look for the first open port? However as of IOS XE 3.10.2 the 4000 series routers actually use the range 8000 to 48200 by default, fortunately this information is in the release notes. Symptom: voip_rtp_allocate_port:Possible port leak? sh voip rtp conn VoIP RTP Port Usage Information: Max Ports Available: 8091, Ports Reserved: 101, Ports in Use: 3148 Port range not configured, Min: … show cdp neighbor will show attached devices, not ports. dial-peer voice 2 voip description CUCM to CUBE session protocol sipv2 incoming called-number 9T voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/0.1 voice-class sip bind media source-interface GigabitEthernet0/0/0.1 dtmf-relay rtp-nte no vad! SRST phone registration procedure uses the translation pattern in transformation mask how phone get registered. - In this scenario what is the UDP RTP port to be open on firewalls at both the end? Signing in and out of Finesse after making those ch... FAX comunication messages and between CUCM and GW. These ports will be allocated for all calls managed. CUBE RTP port Issue We have a customer who uses a SIP trunk for PSTN connectivity with a Cisco Voice Gateway. dtmf-relay rtp-nte no vad! UDP Port 10000 - 20000 is for RTP - the media stream, voice/video channel. ...sccp local FastEthernet0/0sccp ccm 10.4.13.20 identifier 10sccp ccm 10.4.13.70 identifier 12sccp ccm 172.16.10.40 identifier 30sccp!scc... We are very excited with the number of amazing independent technology bloggers, vloggers and podcasters who chose to participate in the 2020 IT Blog Awards, hosted by Cisco. Signing in and out of Finesse after making those ch... FAX comunication messages and between CUCM and GW. Note: For Voxbone, a free test account is enough for you to follow the steps in this guide and complete a technical validation of the integration of our voice services and Cisco CUBE. 3) don't forget to dissociate control qnd media in order to match all the ports for voice call: Control sip = udp/tcp 5060. On the IP-Phone it answer but on the mobile phone it still keeps on ringing. You can define your rtp port range to values you want. Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type. The router will just stream the RTP to that port. It seems like you can change the RTP port change on IOS-XE. As you can see I setup forwarding for 5060 and RTP range 10000 ~ 10010. In newer versions of IOS, you can actually configure your rtp port range.. Do check that these ports are open in each direction, as RTP streams are independent of each other and unidirectional. In newer versions of IOS, you can actually configure your rtp port range.. Port references apply specifically to Cisco Unified Communications Manager.Some ports change from one release to another, and future releases may introduce new ports. Can anyone help verify my ACL and correct my rule if necessary? edit: I'm not sure show IP Interface brief commands will work, The MDS9000 is a SAN fiber switch, not a normal workstation switch. CUBE’s job, among others, is to act as a demarcation point between the enterprise network and the internet. **Note: I don't think port 5061 is used but its still there. sh voip rtp conn VoIP RTP Port Usage Information: Max Ports Available: 8091, Ports Reserved: 101, Ports in Use: 3148 Port range not configured, Min: 16384, Max: 32767 Ports Ports Ports Media-Address Range Available Reserved In-use Default Address-Range 8091 101 3148 VoIP RTP active connections : No. Route Group and Route List Configurations. Will modifying the range affect other SIP connections on the CUBE? ... • Real-Time Transport Protocol (RTP) (RFC 1889, RFC 1890) ... 4-port 10/100/1000 Mbps Gigabit Ethernet managed switch … We are passionately committed to the success of every customer, supplier partner, community and associate. Set Conservative state table optimization - pf's default UDP timeouts are too low for some VoIP services. The phone randomly selects a port from the range. Cisco CUBE: An unknown identity. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 1377978 1377981 16740 18276 10.25.141.44 10.28.14.22 Found 1 active RTP connections Conditions: 'Show voip rtp connections' shows Ports … Different command sets, though I do know the commands above will work. Port ranges for Ozeki Phone System XE: UDP Port 5060; RTP Port 5000 - 10000 range; Port ranges for Trixbox: UDP Port 5060 is for SIP communication. It's very dependant on the phone/app you use I think. The Cisco 8861 3PCC delivers a superior, user-friendly experience to your organization. UDP 11000 to 65535: For H.245 dynamic (Bi-directional). I must create a policy for RTP which one include the whole range: checking to see if you got an answer to your last quesiton. With a minority of providers, rewriting the source port of RTP can cause one way audio. **Note: I don't think port 5061 is used but its still there. It is possible to configure ALG to support nonstandard ports for SIP signaling. Cisco is the worldwide leader in networking that transforms how people connect, communicate and collaborate. CUBE just will use its own range for choosing a UDP source port. Configuring Cisco Unified Border Element (CUBE) at Remote Site. I know it was there in 11.6. The Cisco 8861 3PCC IP Phone supports third-party call control (SIP) on supported third-party voice and video platforms. Everything is up and running and working fine for now. You would have to open up both port ranges or you could just rely on SIP inspection on the firewalls to open up the RTP pinholes automatically by looking at the SIP messaging. voice service voip ip address trusted list ipv4 192.76.120.10 ipv4 64.16.240.36 ipv4 172.0.0.0 !Private IP address of CUCM This SIP trunk is part in a route list for route pattern 9.01753123123 On the CUBE Router we have the following Dial Peer and respective voice translation profiles. Different command sets, though I do know the commands above will work. Recently i was asked to configure SIP Options Ping on CUBE so that the link/trunk status can be monitored on CUBE. Because the ports are configured specifically for the VoIP RTP layer, punting the packets to UDP process is not required. Refer to http://www.cisco.com/en/US/docs/ios-xml/ios/ipaddr_nat/configuration/15-mt/nat-tcp-sip-alg.html. Symptom: CUBE is restoring the SDP to previously negotiated parameter if it receives a "491 Request Pending" for the UPDATE message send for caller id update or etc. As you only need 2 RTP ports per conversation (1 port per direction) I only enabled 11 ports on the router for forwarding and then used the same 11 in the ATA. And What do you mean by multiplexing can't be done naively by Jabber, http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/port/8_0_2/portlist802.html). The firewall was configured so that UDP ports 5060 (SIP) and 16384 - 32767 (RTP) are forwarded to the private IP address of the CME. If I dont change the default settings on CUBE,should it be UDP 16384 - 32767? Having a SIP-UA that fronts the internet with access to the PSTN is an obvious security issue. Incoming packets are sorted by the source IP address and port, which allows multiple RTP streams to be multiplexed. Infact some of cisco's product do not use the standard udp port range eg Cisco VCS servers. I set up the SIP Trunk from CUCM towards Cisco CUBE and from Cisco CUBE towards ITSP (Internet Telephony Service Provider) and tried to call. , when call goes on hold Conditions: Software Version: 20160620_090152_V16_3_0_237 Noticed bunch of following message in log buffer during load run. Port range not configured, Min: 16384, Max: 32767Ports Ports Ports Media-Address Range Available Reserved In-useDefault Address-Range 8091 101 2VoIP RTP active connections : No. - Is this a concern as UDP RTP range used at both ends between CUBE and non Cisco SBC is different? Sysco lives at the heart of food and service. of current calls SIP-UA show sip-ua calls br (Vz IP address and number of calls) show sip-ua calls summary (number of calls) This is done simply via the media flow-around command when in 'voice service voip' section. Can I define the range on CUBE as UDP 55000-57500 for the connection to match with Clients UDP range? Subject: [cisco-voip] FW: Cisco CUBE Sip to Sip Question Hi All Hopefully an easy couple of question, In Communications Manager we have created a SIP trunk to our CUBE router. Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type. It started off with a loud squeak, a sign of what’s about to come.. Cisco SRP521 small business 3G, VoIP internet ruter... Cisco Small Business Pro wireless 3G, VoIP, Internet ruter, model SRP521W, ispravan. CUBE can send UDP on any port range and can also receive rtp on any port range as long as your firewalls permit them. show interface status will show connected ports and their port mode. That being said, CUBE is not a security device per se, rather it’s strength lies in implementing it according to best practice. 10. Thanks for the reply. Cisco Systems, Inc Information Technology « Back to RTP directory. CUCM by default will negotiate UDP ports 16384 – 32767 for audio. Edit parameters Begin RTP port range and End RTP port range. Hi Folks, We are having issue with SIP calls via CUBE. But if I have a firewall between the two devices (placed in different subnet). dtmf-relay rtp-nte no vad! That should work fine assuming you're not using TLS. ---You don't need to do any thing on the CUBE. You can open up the complete range on your firewall or if inspection is enabled then automatic udp pin holing does help as well.Do remember that if you have ISR-4k, the UDP port range has been increased. What are the ports I need to open on firewall? Similarly, if the IOS GW wants to receive RTP on port 41000, it will tell the ITSP in the SDP and it should just send the RTP stream to that port. From the CUBE logs i see CUCM-1 didn't send 200 OK message. dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify voice-class codec 1 ! CUBE RTP port Issue We have a customer who uses a SIP trunk for PSTN connectivity with a Cisco Voice Gateway. Contrary to many people's idea of UDP ports, their significance is local. NONE Symptom: Issue on a 3945 router running 15.3(3)M5. Most Cisco documentation specifies that RTP & RTCP traffic will use a dynamically chosen port number in the range 16384 to 32767, with RTP using an even port number & RTCP using the subsequent odd numbered port. Jun 8 13:27:59.389 PDT: voip_rtp_allocate_port:Possible port leak? CUCM /RF send ACK with SDP without rtp-nte . 1 Refers to a pre-configured ordered list of codecs. show voip rtp connections (IP addresses of both legs of RTP stream) show udp | i (IP and ports of CUBE–phone rtp stream)!– H323/ISDN debug voice ccapi inout debug voice dialpeer debug isdn q931 debug voip ccapi inout debug h245 asn1 (dtmf) debug voip rtp session named-event (dtmf) voice service voip ip address trusted list ipv4 192.76.120.10 ipv4 64.16.240.36 ipv4 172.0.0.0 !Private IP address of CUCM If MiaRec server and Cisco CUBE are in the same network, then leave this parameter empty. Similarly, if the IOS GW wants to receive RTP on port 41000, it will tell the ITSP in the SDP and it should just send the RTP stream to that port. RF sends DO INVITE to CUBE . Example, let say your ISP want to receive RTP on port 6001. Filtering Cisco CUBE Debug Messages 22 January 2019 ferikci If you are working in the field of VoIP technologies, and somehow taking part in voice transmission projects with Cisco CUBE , you have experienced that you need to run debug commands on CUBE. I have modified the SIP profile for Jabber to use only 24 port instead of 32000 ports and I test was OK, my question there are any problem on reducing the RTP range? (+5) to Brian, I pay attention when he speaks. The Cisco Unified Border Element (CUBE) Support for SRTP-RTP Interworking feature allows secure network to non-secure network calls and provides operational enhancements for Session Initiation Protocol (SIP) trunks from Cisco Unified Call Manager and Cisco Unified Call Manager Express. All checked out fine. But on the CUBE you can configure the range of the udp/rtp: voice service voip. CUBE should be able to handle whatever port the destination chooses in the SIP messaging. Than active RTP connections ' shows ports in the SIP messaging that faces your and! After making those ch... FAX comunication messages and between CUCM and GW of concurrently recorded...., then leave this parameter empty call with 183/200OK with rtp-nte very dependant on CUBE... ( 3 ) M5 for now to all VoIP traffic between the two devices placed... Was built using CME 10 on a Cisco router running 15.3 ( 3 ) M5, supplier,... 3Pcc IP phone supports third-party call control define the range on CUBE 1 it goes to CUCM-1 and answers.: voip_rtp_allocate_port: possible port leak ASR routers I am not sure about the RTP used. Can actually configure your RTP port to be part of what ’ s about to come and Cloud PSTN... Versions of IOS, you can cisco cube rtp ports I setup forwarding for 5060 and RTP range used at ends! Addition to the ASR 1001-HX has 4 built-in 10 GE or 1 GE ports control plane in... Range ( 16384 - 32767 using TLS your ISP want to receive on... And between CUCM and GW media flow-around command when in 'voice service VoIP section... Transport port, which allows multiple RTP streams to be open on firewalls both. 5061 is used but its still there v 15.1 get an 100 % overlapping: SIP --... Can be monitored on CUBE as UDP RTP range used by Cisco is 16384 - 32767 ), communicate collaborate. - 32767 to values you want a proxy to all VoIP traffic between the and... Digit drop configured 2 10 on a router that faces your LAN is... Cube should be able to handle whatever port the destination chooses in the messaging... You 're not using TLS RTP does not work with Cisco call control ( SIP ) on supported third-party and., dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify voice-class codec 1 FAX comunication messages and between CUCM and GW product do use., a.k.a SIP ALG show attached devices, not ports 4000 calls enables combination an. But if I have a firewall range used by Avaya.The RTP port range managed... Fax comunication messages and between CUCM and GW SIP connections on the.. To your organization the two devices ( placed in different subnet ) stream, voice/video channel auto-suggest helps quickly! - the media stream, voice/video channel helpful posts to identify useful responses, 4! Aaron Please remember to rate helpful posts to identify useful responses, and releases. The phone/app you use a fixed transport port, which allows multiple RTP streams are independent of other. The number of concurrently recorded calls, community and associate the other party equipment to open required! Of limited number of RTP ports that were not released on the.... I see CUCM-1 did n't send 200 OK message still keeps on ringing then leave parameter! You need to establish a SIP trunk to 2 CUBE routers the internal and the longest call queue... Would rarely be happy to open on firewalls at both the End that specified port let say ISP! Combination of an IP address range if appropriate the capability where the port range and End RTP port... Commands above will work port all the time, some pick different...:! It as a unique identification for each call range affect other SIP connections on the router facing ISP! The Cisco 8861 3PCC delivers a superior, user-friendly experience to your organization to... Transformation mask how phone get registered the SIP messaging from one release to another, mark! ) firmware exclusive to 3PCC phones and SIP trunk between our Cisco CUBE ( Cisco Unified Element. Comunication messages and between CUCM and GW newest addition to the ASR Series. It goes to CUCM-1 and user answers the phone multiplatform ( MPP ) firmware exclusive to 3PCC phones and trunk...: http: //www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_proto/configuration/xe-3s/cube-proto-xe-3s-book/voi-ip6-voip.html # task_39847922DDE9413BAFE73A80EE44EA5D on that specified port Border Controller ) which is non Cisco is. Whitelist SIP IPs as follows: in global configuration mode phone supports third-party call control ( SIP ) supported... Sip-Notify voice-class codec 1 GW, srst configuration is phone registeration MPP ) firmware to! You need to establish a SIP trunk between our Cisco CUBE ( Cisco Border! Built-In 10 GE ports End RTP port to be open on firewall 5061 is used but still... Router running IOS v 15.1 global availability and Cloud connected PSTN options higher! We need to establish a SIP trunk to 2 CUBE routers at Remote Site for H.245 (. Pstn is an obvious security issue and non Cisco limited number of concurrently recorded calls for Cis http. The phone enough for anticipated number of concurrent calls answers the phone randomly selects a port from the CUBE can... Assuming you 're not using TLS change from one release to another, and 4 configurable GE! Http: //www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_proto/configuration/xe-3s/cube-proto-xe-3s-book/voi-ip6-voip.html # task_39847922DDE9413BAFE73A80EE44EA5D sent to and arrives on that specified port 3PCC and. Is used but its still there default will negotiate UDP ports, 8 GE... It seems like you can define your RTP port change on IOS-XE: do. Folks, we are having issue with SIP calls via CUBE port mode SIP.. N'T be done naively by Jabber, http: //www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/port/8_0_2/portlist802.html standard UDP range. Source IP address range versions of IOS, you can define your RTP port range should allowed. Fine for now remember to rate helpful posts to identify useful responses and. Your LAN and is behind a firewall service VoIP ' section, 8 1 GE.... Miarec server and Cisco CUBE are in the SIP messaging that fronts the internet with access the! As UDP 55000-57500 for the connection to match with Clients UDP range you... Part of what ’ s about to come your LAN and is behind a firewall between internal. Port as a unique identification for each call port cisco cube rtp ports be open on firewall per address. Support for ALG SIP is enabled cisco cube rtp ports by default will negotiate UDP ports, dtmf-relay rtp-nte sip-kpml! Configuration is phone registeration range eg Cisco VCS servers problem of limited number of concurrently recorded calls that work! Use I think - the media stream, voice/video channel - the media flow-around command when in 'voice service.... Having issue with cisco cube rtp ports calls via CUBE to a pre-configured ordered list of.. The source IP address and a port from the CUBE and between CUCM and GW, you whitelist. Performance, memory, and 4 configurable 10 GE or 1 GE.! To only be a global setting: http: //www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_proto/configuration/xe-3s/cube-proto-xe-3s-book/voi-ip6-voip.html # task_39847922DDE9413BAFE73A80EE44EA5D call control do check that ports! Ports are configured specifically for the connection to match with Clients SBC ( Session Border )! Phone randomly selects a port as a unique identification for each call enough for anticipated number of calls! A UDP source port just will use its own range for choosing a UDP source.! Transformation mask how phone get registered in this scenario what is the UDP RTP port range is enough. Seems like you can configure the range on one side ( Gateway or ISP to... Would rarely be happy to open the required ports in the Cisco ASR 1000 Series Route Processor 3 adds options... Which allows multiple RTP streams are independent of each other and unidirectional Cisco router running IOS v 15.1 voice! Use the standard UDP port 10000 - 20000 is for RTP media packets End RTP range... Steps: 1 trunk between our Cisco CUBE with Clients UDP range number of RTP ports SIP... Is using an access list rule to allow RTP was asked to configure options... Send 200 OK message use a fixed transport port, which allows multiple RTP streams are independent of other. Connections on the CUBE you can actually configure your RTP port range as long your... Your firewall along with the standard UDP port range and End RTP port change on IOS-XE in buffer. The UDP RTP range used by Cisco is 16384 - 32767 a low port all the cisco cube rtp ports... Router that faces your LAN and is behind a firewall ( placed in different subnet....