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marco_81
Beginner

Cisco Webex devices Calling - network requirements

Hi all,

i need to understand some points about Cisco Webex device calling and network requirements. The deployment i'm focused on is only made of Cisco Webex devices registering to the cloud (no Teams apps).

These devices make intra-customer, intra-site calls, they also can make outbound audio\video calls to join Cisco Webex meetings hosted on other domains.

On my edge firewall i'd open only outbound connection to destinations documented here:

https://help.webex.com/en-us/b2exve/Port-Reference-Information-for-Cisco-Webex-Calling

the destination ports in my case are:

TCP\8934                   signalling to webex SIP-TLS

UDP\19560-65535     media to webex  SRTP

TCP\80, 443               firmware management

UDP\123                     NTP

TCP-UDP\53               DNS

 

No Inbound connection from the internet to internal network.

 

To register devices to Webex Cloud i need a minimum version of CE firmware on each device, do 80-443 ports manage this stage?

If devices from the internal network go directly to the cloud, do they apply SIP ALG or similar functionality on their own?

 

thanks for help

 

UPDATE:

finally i found the right document where it is explained:

With the Cisco Webex Calling product, the challenges presented by the presence of a NAT are addressed. A technique called NAT Traversal is used to overcome the issues created by the presence of a NAT. Part of the Cisco Webex Calling call control platform is responsible for maintaining constant communication with all SIP devices. This constant communication ensures that the NAT bind timer never expires, effectively making the dynamic bind permanent. Without this, a SIP device in a private network would not be able to receive calls. Also, the Cisco Webex Calling call control platform uses a technique called Media Relay to overcome the issue where the NAT does not manipulate application layer information. This functionality allows the call control platform to discover the public IP address and port of the RTP stream once the SIP device sends out its first RTP packet. The call control platform performs this function on both ends of a call and bridges the two legs of the call together, effectively relaying the traffic from one device to another.

Source: (https://callinghelp.webex.com/wp-content/uploads/2019/05/WC-Customer-Network-Minimum-Requirements-Guide-v2.2_062019.pdf)

1 REPLY 1
marco_81
Beginner

Nobody has idea how would device change SDP information in SIP messages when outbound connection traversing NAT is going to the Webex cloud?  

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