Asterisk 20.8.1 pjsip in private AKS

Asterisk Asterisk WebRTC
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Asterisk 20.8.1 pjsip in private AKS

We have an one asterisk pod in private Azure kubernetes. We’ve set hostnetwork: true to use node rtp ports from node(10000-20000).

Our network contains from vnet /16 and subnet /17. Kubernetes is inside vnet > subnet.
There are 2 Network Security Group(NSG):

  1. vnet NSG;
  2. default kubernetes resource group NSG.
    In both NSG configured port forward necessary ports: 5060/5061, 8088/8089 and 10000-20000.

For external access we use Azure public IP which mapped to asterisk as service type LoadBalancer.

A transport configuration:

local_net=10.0.0.0/16(vnet cidr)
external_media_address=Azure Public IP
external_signaling_address=Azure Public IP

An endpoint configuration:

rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes

If test a call via tls (microsip) all work. But if use wss there are a problems: no audio, also calls sometimes interrupt after 30 sec etc.

I understand the asterisk is behind NAT. We have the same configuration in microk8s locally only without load balancer. As external IP we use a corp router’s IP and to do port forward to it. All work as well.

What I have to configure else?

I’ve hear about Kamailio because cloud LB have troubles with rtp ports and the kamailio can fix these problems. But I didn’t found good example how to configure aster + k8s + kamailio.

I wanted to attach voip call log, but new users can’t. How can I send a log file?

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