[OpenSIPS-Users] LB + RTPProxy

[Razvan Crainea]

Hi, Vladimir!

Please provide us a complete SIP trace of the call. You can send me on 
private if you want.

Regards,

--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com


On 04/20/2012 05:05 PM, Vladimir Vakulov wrote:
>
> 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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