Hi Steven,
looking at your config it doesn't seems you are testing for nat in your main
route. only on reply route your have nat_uac_test function called.you need to
do the same in main route and set the NAT flag otherwise your condition "if (
isflagset(NAT) ) {rtpproxy_offer("of", "OPENSIPS IP");}" to use rtpproxy won't
work. you can verfiy it by adding an xlog statement inside that condition. As
per the given config it shouldn't print anything in log.
On Tuesday, 9 October 2018, 10:55:02 PM GMT+5:30, Steven Platt
<[email protected]> wrote:
Good morning,
I have an installation of OpenSIPS 2.3.5, with RTPProxy running on a single
server. RTP Proxy is running as normal, and logs show support for it enabled
during initial connection leg.
My error is that Opensips does not update the connection IP (c=) of the SDP to
force media to be proxied with RTPProxy. Instead, it keep the endpoint IP,
which is behind a NAT, because of this - I have no audio.
Is there something I miss in the configuration to enforce the update of the
connection IP in the SDP? (so that media goes through opensips/rtpproxy)
----------------------------
My flow:
desktop client (zoiper) <--> corporate NAT <--> OPENSIPS <--> carrier NAT <-->
android (zoiper)
Invite SDP Sent from Desktop Zoiper Client:
Via: SIP/2.0/TCP [CORPORATE
NAT]:59401;branch=z9hG4bK-524287-1---fecce2d50d9d5c20;rportMax-Forwards:
70Contact: <sip:1000@[CORPORATE NAT]:59401;transport=TCP>To:
<sip:1001@[OPENSIPS]:5060;transport=TCP>From:
<sip:1000@[OPENSIPS]:5060;transport=TCP>;tag=b27a0843Call-ID:
QMoyxf6JGTFYvxS5X8NsnA..CSeq: 2 INVITEAllow: INVITE, ACK, CANCEL, BYE, NOTIFY,
REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBEContent-Type:
application/sdpProxy-Authorization: Digest username="1000",realm="[OPENSIPS
IP]",nonce="5bbcdde1000000172b9f0086711cd36194c50f208fa420de",uri="sip:1001@[OPENSIPS
IP]:5060;transport=TCP",response="a609cb9d82930d2d32668d8d51d64cb4",algorithm=MD5User-Agent:
Z 5.2.19 rv2.8.99Allow-Events: presence, kpml, talkContent-Length: 161
v=0o=Z 0 0 IN IP4 [DESKTOP IP]s=Zc=IN IP4 [DESKTOP IP]t=0 0m=audio 8000 RTP/AVP
0 101 8a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=sendrecv
The 200OK sent by Opensips to the calling device:
Via: SIP/2.0/TCP [CORPORATE NAT] :59401;received=[CARRIER
IP];branch=z9hG4bK-524287-1---fecce2d50d9d5c20;rport=59401Record-Route:
<sip:[OPENSIPS IP];transport=tcp;lr;did=081.1ad6d9>Contact:
<sip:1001@[CORPORATE NAT]:50758;transport=TCP>To: <sip:1001@[OPENSIPS
IP]:5060;transport=TCP>;tag=07be6967From: <sip:1000@[OPENSIPS
IP]:5060;transport=TCP>;tag=b27a0843Call-ID: QMoyxf6JGTFYvxS5X8NsnA..CSeq: 2
INVITEAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBEContent-Type: application/sdpUser-Agent: Zoiper rv2.8.105Allow-Events:
presence, kpml, talkContent-Length: 245
v=0o=Zoiper 0 1 IN IP4 [ANDROID IP]s=Zoiperc=IN IP4 [ANDROID IP]t=0 0m=audio
42032 RTP/AVP 0 3 8 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:8
PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=sendrecv
---------------------------------------------------
OpenSIPS Config
route[relay] { if (is_method("INVITE")) {
if ( isflagset(NAT) ) {
rtpproxy_offer("of", "OPENSIPS IP"); }
t_on_branch("per_branch_ops");
t_on_reply("handle_nat"); t_on_failure("missed_call"); }
if (isflagset(NAT)) { add_rr_param(";nat=yes");
}
if (!t_relay()) { send_reply("500","Internal Error");
}; exit;}
onreply_route[handle_nat] {
if (nat_uac_test("1"))
fix_nated_contact(); if ( isflagset(NAT) )
rtpproxy_answer("of", "OPENSIPS IP"); xlog("incoming reply\n");}
----------------------------------------------
I also do not see the (";nat=yes") being added in the SDP. Do I understand
correct that the script is not catching this call and flagging it correct as
NAT?
At this time, all signaling works as normal - only media is not being pinned to
the opensips IP in the 200 OK response.
Thanks in advance for any guidance on this one.
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
