I've made some changes and now I'm able to get audio to rtpengine but still 
can't get it to the browser, I'm using flag "WebRTC"

Logs:
index="1" port="11352"]: [ice] ICE negotiated: peer for component 1 is 
10.5.0.2:22438
index="1" port="11352"]: [ice] ICE negotiated: local interface 10.5.0.8
index="1" port="11352"]: [crypto] DTLS: Peer certificate accepted
index="1" port="11352"]: [crypto] DTLS-SRTP successfully negotiated using 
AES_CM_128_HMAC_SHA1_80
index="1" port="11352"]: [crypto] DTLS-SRTP successfully negotiated using 
AES_CM_128_HMAC_SHA1_80
index="1" port="11352"]: [core] Confirmed peer address as 10.5.0.2:22438
index="1" port="11352"]: [srtp] SRTP output wanted, but no crypto suite was 
negotiated
index="1" port="11352"]: [rtcp] SRTCP output wanted, but no crypto suite was 
negotiated
...
[core] Closing call due to timeout
[core] Final packet stats:
[core] --- Tag '7eiernq2ig', created 1:51 ago for branch ''
[core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec
[core] --------- Port        10.5.0.8:14934 <>  176.223.17.217:55146, SSRC 0, 
in 0 p, 0 b, 456 e, 111 ts, out 0 p, 0 b, 0 e
[core] --- Tag 'c73ccc40-a836-4ac7-8968-0d84f7d40b67', created 1:51 ago for 
branch ''
[core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using G722/8000
[core] --------- Port        10.5.0.8:11352 <>        10.5.0.2:22438, SSRC 
221a786, in 456 p, 41900 b, 0 e, 98 ts, out 7 p, 1946 b, 0 e
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