[SR-Users] Kamailio/RTPengine as a proxy for FreePBX/Asterisk...

Sergiu Pojoga pojogas at gmail.com
Wed Sep 29 07:34:44 CEST 2021 

RFC3327

On Wed., Sep. 29, 2021, 12:50 a.m. Micah Quinn, <micah.quinn at sipiq.com>
wrote:

> Thank you for looking at this Sergiu.
>
> Yes, you are correct that Asterisk's INVITE to the receiving extension is
> bouncing between 10.252.1.14 and 192.168.123.10.
>
> Based on the above, how do you expect this call to reach the softphone at
> 10.0.0.142?
>
> I suppose that is the question at issue. And you'll have to forgive me for
> any ignorance on the details of SIP; I'm still learning.
>
>    - Should I be adding a Via header to the message?
>    - Should I be modifying the Contact header? (BTW, I'm already
>    modifying the contact header to point to 10.252.1.14. If I don't, the
>    endpoint shows offline/unavailable. As I understand it, Contact header has
>    nothing to do with message routing.)
>    - Should I be saving the location of the endpoint in Kamailio and
>    doing a lookup on inbound messages?
>
> Given that I've been assured that this is a common use case for Kamailio,
> I have to admit that all of the howto's and example .cfg files I've read
> and tried do not seem to completely address this situation.
> ------------------------------
> *From:* sr-users <sr-users-bounces at lists.kamailio.org> on behalf of
> Sergiu Pojoga <pojogas at gmail.com>
> *Sent:* Tuesday, September 28, 2021 9:25 AM
> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Subject:* Re: [SR-Users] Kamailio/RTPengine as a proxy for
> FreePBX/Asterisk...
>
> This says it all:
>
> 2021/09/28 04:45:07.358826 192.168.123.10:7330 -> 10.252.1.14:5060
> INVITE sip:1093 at 10.252.1.14 SIP/2.0
>
> Based on the above, how do you expect this call to reach the softphone at
> 10.0.0.142?
>
> Also, it's pretty easy to see from the provided traces that the call is
> bounding back and forth between 10.252.1.14 <=> 192.168.123.10, never
> reaching the softphone.
>
> You have a rather long way to making this scenario work, which is good,
> you'll get to learn a few new things as an ITSP.
>
> Good luck.
>
> On Tue, Sep 28, 2021 at 1:00 AM Micah Quinn <micah.quinn at sipiq.com> wrote:
>
> OK, then some more details and some questions. My network configuration
> is as follows:
>
> 10.0.0.142             10.0.0.200               10.252.1.14
>     10.252.1.1    192.168.123.5            192.168.123.10
> [softphone]   <-------->  [kamailio/rtpengine]  <---------VPN--------->
> [VPN server] <------------------> [FreePBX}
>
> There is no NAT'ing involved/enabled. I'm running RTPEngine on the same
> machine as Kamailio.
>
> With my current configuration I can call the PBX directly without issue.
> (i.e. access my voicemail, IVRs, conference rooms, etc.). However, I can
> still not make an extension-to-extension call. Asterisk responds to the
> INVITE with a "401 Unauthorized" message.I have two extensions registered
> (1093 and 10931):
>
>  Endpoint:  1093/1093                                            Not in
> use    0 of inf
>      InAuth:  1093-auth/1093
>         Aor:  1093                                              10
>       Contact:  1093/sip:1093 at 10.252.1.14                  a49a850887
> Avail        85.409
>
>  Endpoint:  10931/10931                                          Not in
> use    0 of inf
>      InAuth:  10931-auth/10931
>         Aor:  10931                                             10
>       Contact:  10931/sip:10931 at 10.252.1.14                3690dfd96d
> Avail        85.225
>
> Below are two packet captures from the Kamailio machine and the Asterisk
> machine. If more information is needed, I'll be happy to supply the
> specifics. Thanks to anyone that's willing to take the time to look this
> over. (Alternatively, if somebody wants to suggest a kamailio.cfg file for
> my specific use case, I'd be happy to test that on my setup as well.)
>
> On the Kamailio machine:
> ---------------------------------------
>
> 2021/09/28 04:45:07.358826 192.168.123.10:7330 -> 10.252.1.14:5060
> INVITE sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 192.168.123.10:5060>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub, histinfo
> Session-Expires: 1800
> Min-SE: 90
> P-Asserted-Identity: "10931" <sip:10931 at 192.168.123.10>
> Max-Forwards: 70
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Type: application/sdp
> Content-Length:   341
>
> v=0
> o=- 585379038 585379038 IN IP4 192.168.123.10
> s=Asterisk
> c=IN IP4 192.168.123.10
> t=0 0
> m=audio 18074 RTP/AVP 0 8 3 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
>
> 2021/09/28 04:45:07.365188 10.252.1.14:5060 -> 192.168.123.10:7330
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e;received=192.168.123.10
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Server: kamailio (5.3.2 (x86_64/linux))
> Content-Length: 0
>
>
>
> 2021/09/28 04:45:07.366400 10.252.1.14:5060 -> 192.168.123.10:5060
> INVITE sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;received=192.168.123.10;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 192.168.123.10:5060>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub, histinfo
> Session-Expires: 1800
> Min-SE: 90
> P-Asserted-Identity: "10931" <sip:10931 at 192.168.123.10>
> Max-Forwards: 69
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Type: application/sdp
> Content-Length:   349
>
> v=0
> o=- 585379038 585379038 IN IP4 10.252.1.14
> s=Asterisk
> c=IN IP4 10.252.1.14
> t=0 0
> m=audio 14618 RTP/AVP 0 8 3 111 9 101
> a=maxptime:150
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
> a=rtcp:14619
> a=ptime:20
>
>
> 2021/09/28 04:45:07.409622 192.168.123.10:5060 -> 10.252.1.14:5060
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 10.252.1.14;rport=19725;received=192.168.123.5;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> CSeq: 13326 INVITE
> WWW-Authenticate: Digest
> realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth"
> Server: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
> 2021/09/28 04:45:07.412926 10.252.1.14:5060 -> 192.168.123.10:5060
> ACK sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 ACK
> Max-Forwards: 69
> Content-Length: 0
>
>
>
> 2021/09/28 04:45:07.413090 10.252.1.14:5060 -> 192.168.123.10:7330
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> CSeq: 13326 INVITE
> WWW-Authenticate: Digest
> realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth"
> Server: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
> 2021/09/28 04:45:07.455640 192.168.123.10:7330 -> 10.252.1.14:5060
> ACK sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 ACK
> Max-Forwards: 70
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
> On the FreePBX machine:
> ---------------------------------------
>
> 2021/09/28 04:45:07.342242 192.168.123.10:5060 -> 10.252.1.14:5060
> INVITE sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 192.168.123.10:5060>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub, histinfo
> Session-Expires: 1800
> Min-SE: 90
> P-Asserted-Identity: "10931" <sip:10931 at 192.168.123.10>
> Max-Forwards: 70
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Type: application/sdp
> Content-Length:   341
>
> v=0
> o=- 585379038 585379038 IN IP4 192.168.123.10
> s=Asterisk
> c=IN IP4 192.168.123.10
> t=0 0
> m=audio 18074 RTP/AVP 0 8 3 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
>
> 2021/09/28 04:45:07.390644 10.252.1.14:5060 -> 192.168.123.10:5060
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e;received=192.168.123.10
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Server: kamailio (5.3.2 (x86_64/linux))
> Content-Length: 0
>
>
>
> 2021/09/28 04:45:07.392235 192.168.123.5:19725 -> 192.168.123.10:5060
> INVITE sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;received=192.168.123.10;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 192.168.123.10:5060>
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 INVITE
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub, histinfo
> Session-Expires: 1800
> Min-SE: 90
> P-Asserted-Identity: "10931" <sip:10931 at 192.168.123.10>
> Max-Forwards: 69
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Type: application/sdp
> Content-Length:   349
>
> v=0
> o=- 585379038 585379038 IN IP4 10.252.1.14
> s=Asterisk
> c=IN IP4 10.252.1.14
> t=0 0
> m=audio 14618 RTP/AVP 0 8 3 111 9 101
> a=maxptime:150
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
> a=rtcp:14619
> a=ptime:20
>
>
> 2021/09/28 04:45:07.393454 192.168.123.10:5060 -> 192.168.123.5:19725
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 10.252.1.14;rport=19725;received=192.168.123.5;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> CSeq: 13326 INVITE
> WWW-Authenticate: Digest
> realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth"
> Server: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
> 2021/09/28 04:45:07.438326 192.168.123.5:19725 -> 192.168.123.10:5060
> ACK sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 ACK
> Max-Forwards: 69
> Content-Length: 0
>
>
>
> 2021/09/28 04:45:07.438558 10.252.1.14:5060 -> 192.168.123.10:5060
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> CSeq: 13326 INVITE
> WWW-Authenticate: Digest
> realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth"
> Server: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
> 2021/09/28 04:45:07.439339 192.168.123.10:5060 -> 10.252.1.14:5060
> ACK sip:1093 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP 192.168.123.10:5060
> ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93
> CSeq: 13326 ACK
> Max-Forwards: 70
> User-Agent: FPBX-16.0.10.27(17.9.4)
> Content-Length:  0
>
>
>
>
>
> ------------------------------
> *From:* Henning Westerholt <hw at skalatan.de>
> *Sent:* Saturday, September 11, 2021 3:16 PM
> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Cc:* Micah Quinn <micah.quinn at sipiq.com>
> *Subject:* RE: Kamailio/RTPengine as a proxy for FreePBX/Asterisk...
>
>
> Hello Micah,
>
>
>
> using Kamailio as front-end/balancer for one or more asterisk instance(s)
> is a classic use case for Kamailio.
>
>
>
> Have a look to the Asterisk log why you get some authentication request,
> probably you need to “tell” Asterisk to trust the Kamailio (IPs).
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users <sr-users-bounces at lists.kamailio.org> *On Behalf Of *Micah
> Quinn
> *Sent:* Friday, September 10, 2021 1:05 AM
> *To:* sr-users at lists.kamailio.org
> *Subject:* [SR-Users] Kamailio/RTPengine as a proxy for
> FreePBX/Asterisk...
>
>
>
> Hello all,
>
>
>
> I'm new to Kamailio, so bear with me as I stumble through this. First,
> I'll describe what I'm trying to achieve at a high level and then perhaps
> somebody can advise me on whether Kamailio is a good fit for this solution
> or not. I'd like to be able to deploy a small appliance type server to our
> customer's sites that just runs Kamailio and a VPN connection back to our
> datacenter. At our datacenter, we run virtualized instances of Asterisk for
> each of our customers. The idea is that Kamailio would act as a transparent
> proxy through to the Asterisk instance under nominal conditions and as a
> basic SIP router in the case that the Asterisk instance is unavailable.
> This degraded functionality would then at least allow extension to
> extension calling even if the Internet or Asterisk instance is down.
>
>
>
> I'm currently using dispatcher with a single entry in preparation for a
> time when we might want to failover to another Asterisk instance. I'm
> forwarding all REGISTER and INVITE messages to the server chosen from
> ds_select_dst. Initially this all seems to work as I can register with a
> softphone and pjsip show endpoints shows my softphone connected. However,
> when I attempt to call any extension (my own or another) Asterisk responds
> to the INVITE message with a "401 Unauthorized" message and the typical
> "The person at extension XXXX is unavailable...".
>
>
>
> I know that more details might be necessary to troubleshoot this, but I
> didn't want to include everything in one post and risk cluttering it up
> with unnecessary information. If anyone can confirm that this is a
> reasonable way to approach the problem, I can then provide whatever
> relevant data is necessary to get deeper into it. (I've used sngrep,
> logging, asterisk cli, etc.)
>
>
>
> Thanks in advance for any help.
>
>
> __________________________________________________________
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> __________________________________________________________
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users at lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
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> Edit mailing list options or unsubscribe:
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