Asterisk no audio issue with no STUN server set on the browser


#1

We have asterisk 15.4 on EC2 and are using sipML5 for placing calls to asterisk through to Bandwidth.com. When we give ice_servers = [] on the browser side with sipML5 (no STUN), we do not hear any audio. The same seems to be working fine with asterisk 15.5 with the same pjsip.conf and rtp.conf configuration(checked release notes and do not see any items related to ICE). In both cases, browser sends private IP in the header package but the latter is working fine.

In the asterisk logs, we only see got and sent rtp packets from bandwidth.com to asterisk - but no packets from the browser. The very first packet exchange on tcpdump, shows SSH, server: encrypted packet from asterisk private IP to browser private IP (in the first instance) and browser public IP (in the second instance where it is working). TLSv1.2 client hello and server hello messages are exchanged with browser’s public IP on both instances.

On the TCP dump, we observe that there are STUN binding request user with private IP but no binding success response (which it is getting on the public IP with asterisk 15.5). As per my understanding, as there is no STUN success response, there is no media flow. Can you please suggest what might be the issue?

This is the pjsip configuration of the browser side on asterisk:

[sipML5]
type = aor
max_contacts = 1

[sipML5]
type = auth
username = sipML5
password = xxxxx

[sipML5]
type = endpoint
context = outgoing
dtmf_mode = none
;rtp_symmetric=yes
disallow = all
allow = ulaw
force_rport=yes
ice_support = yes
direct_media = no
use_avpf = yes
auth = sipML5
outbound_auth = sipML5
media_encryption=dtls
;use_received_transport=yes
media_use_received_transport=yes
dtls_verify = fingerprint
dtls_cert_file = xxxx
dtls_ca_file = xxxx
dtls_setup = actpass
dtls_cipher=ALL
dtls_private_key=xxxx
rtcp_mux=yes
aors = sipML5