[OpenSIPS-Users] LB + RTPProxy

[Vladimir Vakulov]

Good afternoon.

For several days I try to construct a scheme LB RTPProxy:
incoming calls -> GW [10.2.42.70] -> [10.2.42.74] OpenSIPs + RTPProxy [192.168.1.2] -> [192.168.1.3 (/ 4/5/6)] 4x Asterisks (in a test using only the 192.168.1.3)
Calls come in Asterisk, but RTP session is established directly between Asterisk andGW. When I add the script to use RTPProxy, it appears only in the INVITE message and RTP session is established between 192.168.1.2 and 192.168.1.3, but not with thecallers. In the "ACK" and "200 OK" everything as it was before.
Example of 200
"
<--- SIP read from UDP: 10.2.42.74:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.42.70:5060; received = 10.2.42.70; branch = z9hG4bK342cdd3c; rport = 5060
Record-Route: <sip:192.168.1.2;r2=on;lr;ftag=as3d53f405;did=0e8.e8c24272>
Record-Route: <sip:10.2.42.74;r2=on;lr;ftag=as3d53f405;did=0e8.e8c24272>
From: "New User" <sip:12345 at 10.2.42.70>; tag = as3d53f405
To: <sip:7441 at 10.2.42.74:5060>; tag = as3b390137
Call-ID: 6e15f78d6bcdbb293620488a251174f8 at 10.2.42.70: 5060
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:7441 at 192.168.1.3:5060>
Content-Type: application / sdp
Content-Length: 255

v = 0,
o = root 1604383785 1604383785 IN IP4 192.168.1.3
s = Asterisk PBX 10.3.0
c = IN IP4 192.168.1.3
t = 0 0
m = audio 18100 RTP / AVP 8 0 3111
a = rtpmap: 8 PCMA/8000
a = rtpmap: 0 PCMU/8000
a = rtpmap: 3 GSM/8000
a = rtpmap: 111 G726-32/8000
a = ptime: 20
a = sendrecv
"

I would be grateful for any help.

Vladimir.

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