Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer.
Here's my network topology:
+---> [asterisk1]
[public_ip] | 10.50.10.131
[router] <---NAT---> [kamailio] <---+
10.50.10.1 10.50.10.120 |
+---> [asterisk2]
10.50.10.132
In my setup i planned to use UAC and DISPATCHER modules. I started from the
"kamailio-basic.cfg" and added some extra lines to handle UAC and
DISPATCHER.
All is working fine when i do a test call from a softphone inside network
10.50.10.0/24.
When a call is coming from the sip carrier, troubles occurs because
asterisk boxes
are sending their internal ip in SDP.
I understand that i need to rewrite SDP in that case, but i actually don't
know how/where.
I've attached kamailio configuration and a sip trace taken with sngrep
where the problem
is visible.
For security reasons, i would like to force the RTP through RTPProxy.
I'm missing something, and need your help me to understand my errors.
Best Regards,
Bruno
Call flow for
929B936-3F1111E5-9C28C7E1-A16E3F0E (Color by Request/Response)
│SIP/2.0 200 OK
80.110.120.10:5060 10.50.10.120:5060
10.50.10.132:5060 │Via: SIP/2.0/UDP
10.50.10.120;branch=z9hG4bKe82d.a3414799f56e1046d9fede67b168a2ae.0;recei
──────────┬───────── ──────────┬─────────
──────────┬───────── │d=10.50.10.120;rport=5060
05:37:29.196212 │ INV (80.110.120.12:57662) │
│ │Via: SIP/2.0/UDP
80.110.120.10:5060;rport=5060;branch=z9hG4bKe82d.0079d4c5.0
│ ──────────────────────────> │
│ │Via: SIP/2.0/UDP
80.110.16.2:5060;rport=61413;received=80.110.16.2;x-route-tag="tgrp:Slot
05:37:29.204187 │ 100 trying -- your call is │
│ │;branch=z9hG4bKA4079B1AD1
│ <────────────────────────── │
│ │Record-Route: <sip:10.50.10.120;lr=on;ftag=32CDDD90-24CE>
05:37:29.205294 │ │ INV (80.110.120.12:57662)
│ │Record-Route:
<sip:80.110.120.10;lr;ftag=32CDDD90-24CE;did=bb31.a983e793>
│ │ ──────────────────────────>
│ │From: <sip:8231288481@80.110.16.2>;tag=32CDDD90-24CE
05:37:29.229975 │ │ 100 Trying
│ │To: <sip:9822147...@voip.carrier.me>;tag=as355bc928
│ │ <──────────────────────────
│ │Call-ID: 929B936-3F1111E5-9C28C7E1-A16E3F0E@80.110.16.2
05:37:29.233990 │ │ 200 (10.50.10.132:10832)
│ │CSeq: 101 INVITE
│ │ <──────────────────────────
│ │Server: Asterisk PBX 13.1-cert2
05:37:29.235113 │ 200 (10.50.10.132:10832) │
│ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAG
│ <────────────────────────── │
│ │Supported: replaces, timer
05:37:29.333226 │ │ 200 (10.50.10.132:10832)
│ │Contact: <sip:9822147941@10.50.10.132:5060>
│ │ <<<────────────────────────
│ │Content-Type: application/sdp
05:37:29.335947 │ 200 (10.50.10.132:10832) │
│ │Content-Length: 347
│ <<<──────────────────────── │
│ │
05:37:29.533537 │ │ 200 (10.50.10.132:10832)
│ │v=0
│ │ <<<────────────────────────
│ │o=root 397373482 397373482 IN IP4 10.50.10.132
05:37:29.535938 │ 200 (10.50.10.132:10832) │
│ │s=Asterisk PBX 13.1-cert2
│ <<<──────────────────────── │
│ │c=IN IP4 10.50.10.132
05:37:29.934272 │ │ 200 (10.50.10.132:10832)
│ │t=0 0
│ │ <<<────────────────────────
│ │m=audio 10832 RTP/AVP 3 18 8 0 101
05:37:29.935421 │ 200 (10.50.10.132:10832) │
│ │a=rtpmap:3 GSM/8000
│ <<<──────────────────────── │
│ │a=rtpmap:18 G729/8000
05:37:30.734051 │ │ 200 (10.50.10.132:10832)
│ │a=fmtp:18 annexb=no
│ │ <<<────────────────────────
│ │a=rtpmap:8 PCMA/8000
05:37:30.735388 │ 200 (10.50.10.132:10832) │
│ │a=rtpmap:0 PCMU/8000
│ <<<──────────────────────── │
│ │a=rtpmap:101 telephone-event/8000
05:37:32.333693 │ │ 200 (10.50.10.132:10832)
│ │a=fmtp:101 0-16
│ │ <<<────────────────────────
│ │a=ptime:20
05:37:32.336746 │ 200 (10.50.10.132:10832) │
│ │a=maxptime:150
│ <<<──────────────────────── │
│ │a=sendrecv
05:37:35.534111 │ │ 200 (10.50.10.132:10832)
│ │
│ │ <<<────────────────────────
│ │
05:37:35.535607 │ 200 (10.50.10.132:10832) │
│ │
│ <<<──────────────────────── │
│ │
kamailio.cfg
Description: Binary data
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