22 Sep
2023
22 Sep
'23
2:58 a.m.
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus