Setting connection limit(call limit) to nodes while load balancing sip traffic over UDP
I have the following setup. SIP call is distributed from SBC to F5 LTM over UDP. Created a SIP persistence profile in LTM using call id as persistence key. Calls are being distributed to all nodes in the round robin and SIP messages are getting persisted. My requirement is to load balance the calls to the least active call node and limit the number of active calls to each node. i.e Nodes shall take only x number of active calls at any time. Active call is a call to which BYE is not received yet. Tried setting connection limit to each node but number of connections is always 1 on the statistics page. The connection limit is always 1 as in my case SIP is over UDP, source and destination IP & Port will be same for all calls. Is there any way in F5 LTM where we can limit the number of calls to each node and load balance the calls to the least call-taking node for SIP over UDP?
F5 LTM and simple centos voice server
i want to load balance 2 centos web servers created pool , pol members with port 5060 created udp profile choose udp for virtual server selected snat automap / in another time created a snat pool connected ip phone to the ip of the virtual server the ip phone registered succesfully but when the first back end server is down or forced down / the phone doesnt register to the second identical one it is stuck on the first
SIP TCP 5061 packets dropped by IP Forwarding Virtual Server
Hello I have a HA F5 LTM Solution running v14.1.2.3 I have an internal vlan (192.168.10.8 as Floating IP) and external vlan (192.168.13.5 as Floating IP) I also have an IP Forwarding Virtual Server that allows all 192.168.10.x servers to send packets through the F5 without making any NAT configuration. So servers in 192.168.10.0/24 network can send all protocol packets through the f5 to reach external networks. For example, 192.168.10.50 server sends a packet to 10.16.75.50 server. First, it reaches 192.168.10.8 (F5 internal vlan), then the packet is sent to a firewall (192.168.13.2 which is in the same vlan as external vlan in f5), then the firewall reaches 10.16.75.50 by static route or because it is one of its local networks. The problem comes when this packets are from SIP protocol (TCP 5061 SIP REGISTER), I see that the f5 drops this packets even though the IP Forwarding Virtual Server should allow them because it allows TCP packets and all ports are permitted. Do I need to create something different from my IP Forwarding virtual server? ps: When I run tcpdump in F5, I see SIP traffic coming to internal vlan interface but nothing egresses by the external vlan interface
Source port of SIP UDP request gets translated regardless of source port setting 'preserve strict'
Hi Guys, I configured a SIP (UDP) service on a 11.5.1 HF3 LTM at a customer site. A SIP profile is bound to the SIP Virtual Service. The LTM ist set up in the L3 Path. Most of the VS configuration ist default. What I see is, that the LTM translates the source port on the traffic from the LTM to the SIP server. The behaviour does not change if I configure 'preserve strict' on the source port setting in the VS: 09:47:50.418275 IP 1.1.1.1.sip > 1.1.2.1.sip: SIP, length: 452 (client to SIP VS on LTM) 09:47:50.418560 IP 1.1.1.1.25350 > 1.1.3.1.sip: SIP, length: 547 (forwarded request to SIP server) 09:47:50.424877 IP 1.1.3.1.sip > 1.1.1.1.25350: SIP, length: 378 (response from SIP server) 09:47:50.424927 IP 1.1.2.1.sip > 1.1.1.1.sip: SIP, length: 277 (response form LTM VS to client) Any ideas? Thanks, RolfPlain old NAT and SIP
Hello, What would be the easiest way to just NAT SIP traffic without terminating SIP connection on F5 as suggested in their official document here https://www.f5.com/pdf/deployment-guides/load-balancing-sip-dg.pdf. I don't want any SIP traffic engineering or anything fancy. Just NAT and SIP adjustment accordingly. I have a SIP gateway and a separate media server (on a different IP). So clients (firewall inside) always go to SIP gateway (firewall outside) and never talk to each other directly. Please advise. Thank You! Bojan
SIP Health Monitor problem
Hi There I am trying to implement UDP SIP Health Monitor and facing with SIP header issue. When F5 sent the SIP Options request, it was sending with F5 FQDN which was resolved to Management IP address. So, when the SIP server reply, it's trying to reply to management IP instead. How do I force F5 to use "Inside" interface IP closed to SIP Server? Thanks in advance! Regards JoeSIP irule for checking dialed string
Hi guys, Please help me with creating na irule for checking a dialed string in SIP INVITE request. I need to drop every call that has !=8 or !=0 symbols in the SIP INVITE. Like this: sip:3286184@206.81.166.10 I need to strip only "3286184" portion, then check if it has 8 or 0 symbols, if not I have to drop the call. That's the irule I've made but it's not working for some reason... when SIP_REQUEST { set meeting_nbr [findstr [SIP::header "INVITE"] ":" 1 "@"] if { not [string length $meeting_nbr] == 8} || { not [string length $meeting_nbr] == 0} { log local0. "SIP Drop [SIP::uri]" drop } } Please help. thanks.
SIP 200 OK dropped when used NAT rules in LTM.
In one of our customer deployment, where BIG-IP LTM is used, sometime we have noticed that for certain SIP MESSAGE packets when 200 OK is received from SIP server, then LTM is dropping the 200 OK when NAT is configured. All the flow is happening over UDP. Can you please indicate any troubleshoot guide by which we can identify the possible reason of packet drop. does LTM has capability to identify SIP 200 OK of SIP MESSAGE only and due to some miss configuration it will get dropped. Please help
Placing Lync 2013 Outbound Calls to a SIP gateway
Hello all. I'm in the process of setting up a simple Lync 2013 system as a small PBX replacement. We have an external SIP trunk provider that serves as our interface to the PSTN. This is a known good trunk/PSTN gateway as Lync can place and receive outside calls with the mediation server's interface exposed directly to the Internet. However, I would like to hide the mediation server behind an F5 BIG-IP LTM for security reasons. I was able to download and install the most recent Lync iApp, and this took care of inbound calls to the mediation server without any issues. However, I can't seem to place outbound calls; I get SIP 503 Service Unavailable errors when I try using a Virtual Server facing the inside interface, with the external PSTN gateway set as a TCP pool member on port 5060. Setting the SIP profile makes no difference, and neither does the network translation setting. Is there anyone familiar with this kind of scenario for Lync 2013? I can provide more details if someone here understands what I'm trying to accomplish. Much appreciated. Gerard