Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[James Hogbin]

That fixed the hangup issue.  Thank you very much.  I’d’ve never spotted that 
in a million years of looking.

Odd that the audio worked at all though

I’m still confused as to why the Teams can route out but the pbx cannot route 
back.  Other than adding the Record route I’m not doing anything different 
between them

if (is_method("INVITE") && !has_totag() && check_source_address(0)) {
xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");
strip(1);
prefix("+44");
record_route_preset("sbc.ip-sentinel.com:5091;transport=tls", 
"137.117.136.143:5060");
add_rr_param(";r2=on");
$rd="sip.pstnhub.microsoft.com";
$rp=5061;
route(relay);
} else if (is_method("INVITE") && ! has_totag() && 
!check_source_address(0)) {
record_route();
xlog("[ROUTE]Incoming call from MS: RURI=$ruri, SI=$si, 
M=$rm\n");
$rd="pbx.ip-sentinel.com";
$rp=5081;
route(relay);

James Hogbin
Director 
 
IP Sentinel 
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com

> On 7 May 2020, at 20:57, Alexey Vasilyev  wrote:
> 
> Ok James,
> 
> Can you please look at your config with more attention:
> 
> if(has_body("application/sdp")){
> xlog("[RTPPROXY] route[relay] we have sdp on this message\n$rm\n");
> rtpproxy_offer("co", "137.117,136.143");
> }
> 
> }
> 
> xlog("[INFO] Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");
> if (!t_relay()) {
> send_reply(500,"Internal Error");
> }
> exit;
> }
> 
> onreply_route[handle_nat] {
> xlog("[INFO] onreply_route[handle_nat]: RR=$rr, RS=$rs, SI=$si\n$mb\n");
> # we receive a reply, we need to check about application/sdp
> # on our body, if we have, we answer that
> if(is_method("ACK") && has_body("application/sdp")){
> xlog("[RTPPROXY] onreply_route[handle_nat] ACK rtpproxy_answer\n");
> rtpproxy_answer("co", "137.117,136.143");
> }
> 
> and change comma to point.
> May be this can help?
> 
> 
> 
> 
> -
> ---
> Alexey Vasilyev
> --
> Sent from: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
> 
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Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[Alexey Vasilyev]

Ok James,

Can you please look at your config with more attention:

if(has_body("application/sdp")){
xlog("[RTPPROXY] route[relay] we have sdp on this message\n$rm\n");
rtpproxy_offer("co", "137.117,136.143");
}

}

xlog("[INFO] Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");
if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}

onreply_route[handle_nat] {
xlog("[INFO] onreply_route[handle_nat]: RR=$rr, RS=$rs, SI=$si\n$mb\n");
# we receive a reply, we need to check about application/sdp
# on our body, if we have, we answer that
if(is_method("ACK") && has_body("application/sdp")){
xlog("[RTPPROXY] onreply_route[handle_nat] ACK rtpproxy_answer\n");
rtpproxy_answer("co", "137.117,136.143");
}

and change comma to point.
May be this can help?




-
---
Alexey Vasilyev
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Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[James Hogbin]

That is not a typo

No clue as to how to change that. The RTPProxy is set up as per the 
documentation. Well as far as I understand it. 

I don’t understand how it works fine from Teams to The PBX 

But not the other way. 

Also how the return path for BYE isn’t working

James Hogbin
Director 
 
IP Sentinel 
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com

> On 7 May 2020, at 19:15, Alexey Vasilyev  wrote:
> 
> Hi James,
> 
> Just to be sure, that this is not a typo:
> 
> check your SDP
> o=  137.117,136.143
> c=IN IP4 137.117,136.143
> 
> This will definitely fail.
> 
> 
> 
> -
> ---
> Alexey Vasilyev
> --
> Sent from: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
> 
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

IP Sentinel Disclaimer 
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delivered and neither it nor any of its contents should be passed on to or used 
by any other person. IP Sentinel Ltd is a limited company registered in England 
and Wales under Registered Number 08648097. Registered Office: Newnhams Wood, 
Horsted Keynes, West Sussex, RH17 7BT. 
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Re: [OpenSIPS-Users] [BLOG] SIP Push Notification with OpenSIPS 3.1 LTS [RFC 8599 support][Part I]

[Slava Bendersky via Users]

WOW, Great Thank you !!! 

volga629 


From: "Liviu Chircu"  
To: "OpenSIPS users mailling list" , "OpenSIPS devel 
mailling list" , busin...@lists.opensips.org, 
n...@lists.opensips.org 
Sent: Thursday, May 7, 2020 3:44:19 PM 
Subject: [OpenSIPS-Users] [BLOG] SIP Push Notification with OpenSIPS 3.1 LTS 
[RFC 8599 support][Part I] 

Hi, folks! 

We now have standards-based SIP Push Notification support in OpenSIPS 
3.1! If you want to learn how they work, as well as how to enable them 
in OpenSIPS, you can follow along this 2-part blog series [1]. Thank 
you and enjoy! 

Stay safe, 

[1]: 
https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/
 

-- 
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www.twitter.com/liviuchircu | www.opensips-solutions.com 


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[OpenSIPS-Users] [BLOG] SIP Push Notification with OpenSIPS 3.1 LTS [RFC 8599 support][Part I]

[Liviu Chircu]

Hi, folks!

We now have standards-based SIP Push Notification support in OpenSIPS 
3.1!  If you want to learn how they work, as well as how to enable them 
in OpenSIPS, you can follow along this 2-part blog series [1].  Thank 
you and enjoy!


Stay safe,

[1]:
https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/

--
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www.twitter.com/liviuchircu | www.opensips-solutions.com


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Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[Johan De Clercq]

Indeed.

On Thu, 7 May 2020, 20:15 Alexey Vasilyev, 
wrote:

> Hi James,
>
> Just to be sure, that this is not a typo:
>
> check your SDP
> o=  137.117,136.143
> c=IN IP4 137.117,136.143
>
> This will definitely fail.
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>
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>
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Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[Alexey Vasilyev]

Hi James,

Just to be sure, that this is not a typo:

check your SDP
o=  137.117,136.143
c=IN IP4 137.117,136.143

This will definitely fail.



-
---
Alexey Vasilyev
--
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Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[Johan De Clercq]

Your contact header looks wrong. Hint: look in the oracle sbc ms teams docs.

On Thu, 7 May 2020, 18:24 James Hogbin,  wrote:

> My set up
>
> Teams <—> sbc.ip-sentinel.com <—> pbx.ip-sentinel.com
>
> I can create a call with audio from Teams -> PBX
> I can hang up that call from Teams but not the PBX
> I cannot route a call from PBX to teams. Although the initial TLS
> handshake does happen
>
> I’m pretty sure it’s something basic to do with headers on ACK & BYE but I
> just cannot spot what I’ve done wrong
>
> This is the initial outbound message from opensips to teams which looks OK
>
> *James Hogbin*
> *Director*
> [image: IP Sentinel Logo] 
> t. +44 (0)20 3011 4150 <+442030114150>
> m. +44 7786910895
> w. https://www.ip-sentinel.com
>
>
> INVITE sip:+448435577...@sip.pstnhub.microsoft.com:5061;transport=tls
> SIP/2.0
> Record-Route:  ;transport=tls;ftag=eect23Z290N9p;lr;r2=on>
> Record-Route: 
> Via: SIP/2.0/TLS 137.117.136.143:5091
> ;branch=z9hG4bK48f8.81967713.0;i=e67586f7
> Via: SIP/2.0/TLS 13.80.245.144:5081
> ;rport=41795;received=10.0.0.4;branch=z9hG4bKac61XQKeXDXHB
> Max-Forwards: 68
> From: "James Hogbin" ;tag=eect23Z290N9p
> To: 
> Call-ID: 0109ad37-0b1f-1239-bdba-000d3aada04e
> CSeq: 19879884 INVITE
> Contact:  ;transport=tls;transport=tls;gw=c6ff36e8-d3de-4fe0-9f1b-9da2888c43a9>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 1339
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "James Hogbin"  >;party=calling;screen=yes;privacy=off
> #015
> v=0
> o=FreeSWITCH 1588835231 1588835232 IN IP4 137.117,136.143
> s=FreeSWITCH
> c=IN IP4 137.117,136.143
> t=0 0
> m=audio 10432 RTP/SAVP 9 0 8 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> [stuff removed]
> a=ptime:20
> m=audio 13446 RTP/AVP 9 0 8 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=nortpproxy:yes
>
> opensips.cfg Source_address(0) is my PBX address=
>
> alias = tls:sbc.ip-sentinel.com:5091
> alias = udp:sbc.ip-sentinel.com:5060
> alias = tcp:sbc.ip-sentinel.com:5060
> advertised_address=137.117.136.143
> listen=udp:10.0.0.5:5060
> listen=tcp:10.0.0.5:5060
> listen=tls:10.0.0.5:5091
>
> ### Routing Logic 
>
> # main request routing logic
>
> route{
> #xlog("L_INFO", "route SIP Message $mb");
> force_rport();
>
> if (!mf_process_maxfwd_header(10)) {
> send_reply(483,"Too Many Hops");
> exit;
> }
>
> if (is_method("OPTIONS")) {
> #xlog("[MS TEAMS] OPTIONS In\n");
> send_reply(200, "OK");
> exit;
> }
>
> # CANCEL processing
> if (is_method("CANCEL")) {
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> # absorb retransmissions, but do not create transaction
> t_check_trans();
>
> if (has_totag()) {
>
> # sequential request within a dialog should
> # take the path determined by record-routing
>
> if(is_method("INVITE") && !check_source_address(0)) {
> xlog("[INFO] In dialog Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");
> t_relay();
> }
> if ( !loose_route() ) {
> # we do record-routing for all our traffic, so we should not
> # receive any sequential requests without Route hdr.
> send_reply(404, "Not here");
> exit;
> }
> # route it out to whatever destination was set by loose_route()
> # in $du (destination URI).
> route(relay);
> exit;
> }
>
> # account only INVITEs
> if (is_method("INVITE")) {
> do_accounting("log");
> }
>
> if (is_method("INVITE") && !has_totag() && check_source_address(0)) {
> xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");
> strip(1);
> prefix("+44");
> record_route_preset("sbc.ip-sentinel.com:5091;transport=tls", "
> 137.117.136.143:5060");
> add_rr_param(";r2=on");
> $rd="sip.pstnhub.microsoft.com";
> $rp=5061;
> route(relay);
> } else if (is_method("INVITE") && ! has_totag() &&
> !check_source_address(0)) {
> record_route();
> xlog("[ROUTE]Incoming call from MS: RURI=$ruri, SI=$si, M=$rm\n");
> $rd="pbx.ip-sentinel.com";
> $rp=5081;
> route(relay);
> }
>
> if (!is_myself("$rd")) {
> append_hf("P-hint: outbound\r\n");
> route(relay);
> }
>
> # requests for my domain
> if (is_method("PUBLISH|SUBSCRIBE")) {
> send_reply(503, "Service Unavailable");
> exit;
> }
>
> if ($rU==NULL) {
> # request with no Username in RURI
> send_reply(484,"Address Incomplete");
> exit;
> }
>
> # do lookup with method filtering
> if (!lookup("location","m")) {
> t_reply(404, "Not Found");
> exit;
> }
>
> # when routing via usrloc, log the missed calls also
> do_accounting("log","missed");
> route(relay);
> }
>
> route[relay] {
> xlog("[INFO] route[relay] SIP Message \n$mb\n");
> # for INVITEs enable some additional helper routes

Re: [OpenSIPS-Users] OpenSIPS as Teams SBC

[James Hogbin]

My set up

Teams <—> sbc.ip-sentinel.com <—> pbx.ip-sentinel.com

I can create a call with audio from Teams -> PBX
I can hang up that call from Teams but not the PBX
I cannot route a call from PBX to teams.  Although the initial TLS handshake 
does happen

I’m pretty sure it’s something basic to do with headers on ACK & BYE but I just 
cannot spot what I’ve done wrong

This is the initial outbound message from opensips to teams which looks OK

James Hogbin
Director 
 
IP Sentinel 
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com

INVITE sip:+448435577...@sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0
 Record-Route: 

 Record-Route: 
 Via: SIP/2.0/TLS 137.117.136.143:5091;branch=z9hG4bK48f8.81967713.0;i=e67586f7
 Via: SIP/2.0/TLS 
13.80.245.144:5081;rport=41795;received=10.0.0.4;branch=z9hG4bKac61XQKeXDXHB
 Max-Forwards: 68
 From: "James Hogbin" ;tag=eect23Z290N9p
 To: 
 Call-ID: 0109ad37-0b1f-1239-bdba-000d3aada04e
 CSeq: 19879884 INVITE
 Contact: 

 User-Agent: FreeSWITCH
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 
REFER, NOTIFY
 Supported: timer, path, replaces
 Allow-Events: talk, hold, conference, refer
 Content-Type: application/sdp
 Content-Disposition: session
 Content-Length: 1339
 X-FS-Support: update_display,send_info
 Remote-Party-ID: "James Hogbin" 
;party=calling;screen=yes;privacy=off
 #015
 v=0
 o=FreeSWITCH 1588835231 1588835232 IN IP4 137.117,136.143
 s=FreeSWITCH
 c=IN IP4 137.117,136.143
 t=0 0
 m=audio 10432 RTP/SAVP 9 0 8 101 13
 a=rtpmap:9 G722/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=rtpmap:13 CN/8000
[stuff removed]
 a=ptime:20
 m=audio 13446 RTP/AVP 9 0 8 101 13
 a=rtpmap:9 G722/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=rtpmap:13 CN/8000
 a=ptime:20
 a=nortpproxy:yes

opensips.cfg Source_address(0) is my PBX address= 

alias = tls:sbc.ip-sentinel.com:5091
alias = udp:sbc.ip-sentinel.com:5060
alias = tcp:sbc.ip-sentinel.com:5060
advertised_address=137.117.136.143
listen=udp:10.0.0.5:5060
listen=tcp:10.0.0.5:5060
listen=tls:10.0.0.5:5091

### Routing Logic 

# main request routing logic

route{
#xlog("L_INFO", "route SIP Message $mb");
force_rport();

if (!mf_process_maxfwd_header(10)) {
send_reply(483,"Too Many Hops");
exit;
}

if (is_method("OPTIONS")) {
#xlog("[MS TEAMS] OPTIONS In\n");
send_reply(200, "OK");
exit;
}

# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
}

# absorb retransmissions, but do not create transaction
t_check_trans();

if (has_totag()) {

# sequential request within a dialog should
# take the path determined by record-routing

if(is_method("INVITE") && !check_source_address(0)) {
xlog("[INFO] In dialog Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");
t_relay();
}
if ( !loose_route() ) {
# we do record-routing for all our traffic, so we should not
# receive any sequential requests without Route hdr.
send_reply(404, "Not here");
exit;
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(relay);
exit;
}

# account only INVITEs
if (is_method("INVITE")) {
do_accounting("log");
}

if (is_method("INVITE") && !has_totag() && check_source_address(0)) {
xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");
strip(1);
prefix("+44");
record_route_preset("sbc.ip-sentinel.com:5091;transport=tls", 
"137.117.136.143:5060");
add_rr_param(";r2=on");
$rd="sip.pstnhub.microsoft.com";
$rp=5061;
route(relay);
} else if (is_method("INVITE") && ! has_totag() && 
!check_source_address(0)) {
record_route();
xlog("[ROUTE]Incoming call from MS: RURI=$ruri, SI=$si, 
M=$rm\n");
$rd="pbx.ip-sentinel.com";
$rp=5081;
route(relay);
}

if (!is_myself("$rd")) {
append_hf("P-hint: outbound\r\n");
route(relay);
}

# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE")) {
send_reply(503, "Service Unavailable");
exit;
}

if ($rU==NULL) {
# request with no Username in RURI
send_reply(484,"Address Incomplete");
exit;
}

# do lookup with method filtering
if 

[OpenSIPS-Users] TLS SNI Checks

[Callum Guy]

Hi All,

Some of our clients are brave enough to access our OpenSIPs WebRTC
gateway using Microsoft Edge.

We've had some teething issues which have been diagnosed as a failed
SNI check due to the character casing, our certificate presents common
and alt names in lowercase (i.e. rtc.opensips.org) however Edge was
checking using a mix of uppercase and lowercase (i.e.
RTC.opensips.org). The server responds with a fatal alert message
"Description: Unrecognized Name (112)" indicating the mismatch.

The fix in our case is just to amend the client configuration however
most other browsers have been demonstrated to fix this by default. I'm
wondering if this is worth patching server side such that the
certificate names and client provided SNI's are held only in lowercase
or a case insensitive match is performed?

Thanks,

Callum

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[OpenSIPS-Users] migration to 2.4

[Miha via Users]

Hello

we are running still on 2.1. Due to some other things I would like first 
to migrate to version 2.4. I went over documentation for version 
migration from 2.1 to 2.2 and from 2.2 to 2.3 and from 2.3. to 2.4.


What I would like to know is what exactly is wrong in my config in where 
i should be looking for.
the main issue is that I do not see this in logs. Log level is 4 (i 
tried aslo with 7 and other leves.)


This are logs:
Is there any other way to find this issue?

  6 21:26:24 debian opensips[6423]: NOTICE:core:main: version: opensips 
2.4.7 (x86_64/linux)
May  6 21:26:24 debian opensips[6423]: INFO:core:main: using 32 Mb of 
shared memory
May  6 21:26:24 debian opensips[6423]: INFO:core:main: using 2 Mb of 
private process memory
May  6 21:26:24 debian opensips[6423]: INFO:core:init_reactor_size: 
reactor size 1024 (using up to 0.03Mb of memory per process)
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:tm:mod_init: TM - 
initializing...
May  6 21:26:24 debian opensips[6423]: INFO:sl:mod_init: Initializing 
StateLess engine
May  6 21:26:24 debian opensips[6423]: NOTICE:signaling:mod_init: 
initializing module ...

May  6 21:26:24 debian opensips[6423]: INFO:rr:mod_init: rr - initializing
May  6 21:26:24 debian opensips[6423]: INFO:maxfwd:mod_init: initializing...
May  6 21:26:24 debian opensips[6423]: INFO:sipmsgops:mod_init: 
initializing...
May  6 21:26:24 debian opensips[6423]: INFO:usrloc:ul_init_locks: locks 
array size 512
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:registrar:mod_init: 
initializing...

May  6 21:26:24 debian opensips[6423]: INFO:acc:mod_init: initializing...
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:mod_init: initializing 
UDP-plain protocol
May  6 21:26:24 debian opensips[6423]: INFO:core:probe_max_sock_buff: 
using rcv buffer of 416 kb
May  6 21:26:24 debian opensips: INFO:core:daemonize: pre-daemon process 
exiting with 0

May  7 09:15:33 debian opensips: NOTICE:core:main: Exiting
May  7 09:15:57 debian opensips: NOTICE:core:main: Exiting
May  7 09:16:56 debian opensips: NOTICE:core:main: Exiting
May  7 09:17:23 debian opensips: NOTICE:core:main: Exiting


thank you
miha
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