date:20200626

[SR-Users] kamailio and rtpengine in docker

2020-06-26 Thread David Villasmil

Hello guys,

I've got a kamailio on a docker and kamailio on another docker on different
boxes.
The issue i'm having is that when kamailio tries to enable rtpengine it
tries pinging it.
The ping arrives at the actual container where rtpengine is running, but
for some reason i haven't figured out, rtpengine (or the container) is not
responding to the ping.

I'm writing here in the hopes that someone has already implemented this an
come across this and a solved (for sure)

I'd appreciate some help.

Many thanks!

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [SR-Users] Kamailio IMS with ENUM

2020-06-26 Thread Valentin Christoph

Hi,

There are two funny things, which I do not understand:

  1.  The first (orig) S-CSCF should already replace the TEL URI by a SIP URI 
and do an ENUM query:

Lines 1140 – 1158 of Kamailio_scscf.cfg:

route[PSTN_handling]
{
 # First, we translate "tel:"-URI's to SIP-URI's:
 # $ru:   tel:+(34)-999-888-777
 # $fu:   sip:t...@foo.com
 # becomes $ru:   sip:+34999888...@foo.com;user=phone

  if (tel2sip("$ru", "$fd", "$ru") < 0)<--THIS SHOULD CONVERT 
TEL URI to SIP URI a)

xlog("L_WARN","Failed to convert $ru to a sip:-URI - M=$rm 
R=$ru F=$fu T=$tu IP=$si:$sp ID=$ci\n\n");

  if ($rU =~ "\+[0-9]+") {

   # Now let's check, if the number can be found in ENUM:

if(!enum_query()) {

 # ENUM failed, send it to the PSTN-Gateway:

 route(PSTN);

 break;

}

  }

}

  1.  Do you have .pcap’s from the traffic between first S-CSCF and I-CSCF? 
What is the Request URI?
You can take detailed logs (#define WITH_DEBUG in scscf.cfg) and analyse them 
-> if necessary, we can help with analysis

1. 2. If the I-CSCF actually receives a TEL URI (what we do not know yet), 
then “why the heck do not we call the enum_query()”? Why do we hit the “604 
Does not exist anywhere”?

b) Let’s keep these questions, until we know, if the S-CSCF has actually sent a 
TEL URI to the I-CSCF in the Request Line of the INVITE.

Lines 537 – 576 of kamailio_icscf.cfg:

route[lir_term_user_unknown]

{

 if (uri =~ "tel:.*") {

   # Let's check, if the number can be found in ENUM:

   if(!enum_query()) {

# ENUM failed, send it to the PSTN-Gateway:

route(PSTN);

break;

   }

   # ENUM resolved to another domain

   if ($rd != NETWORKNAME) {

t_on_reply("initial_request_reply");

t_on_failure("initial_request_failure");

if (!t_relay()) {

 t_reply("500","Error forwarding to external domain");

 exit;

};

exit;

   } else {

t_reply("604","Does not exist anywhere - HSS User Unknown");

exit;

   };

 } else {

   # we received a request for our domain (non-tel), but HSS said "User 
Unknown"

   if ($rd != NETWORKNAME) {

t_reply("604","Does not exist anywhere - HSS User Unknown");

exit;

   } else {

# try to forward non-tel request to other domain

t_on_reply("initial_request_reply");

t_on_failure("initial_request_failure");

if (!t_relay()) {

 t_reply("500","Error forwarding to external domain");

 exit;

};

exit;

   };

 };

}

Hope it helps

Christoph

-Ursprüngliche Nachricht-
Von: sr-users  Im Auftrag von BALL SUN
Gesendet: Donnerstag, 25. Juni 2020 04:29
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] Kamailio IMS with ENUM

Hello

I may not describe the problem clearly. When a UE call a number to the Kamailio 
IMS core, says, the number is 12345678, the R-URI in the SIP INVITE is 
"sip:12345678;phone-context=ims.mnc001.mcc001.3gppnetwork@ims.mnc001.mcc001.3gppnetwork.org;user=phone",

Kamailio P-CSCF pass to INVITE to S-CSCF, and then S-CSCF pass to I-CSCF, 
I-SCSF trigger LIR to HSS with Public-Identity = tel:12345678, This IMPU did 
not provision in HSS, so HSS response  with "DIAMATER_ERROR_USER_UNKNOWN".

On receiving this LIA, I-CSCF return to S-CSCF with "604 Does not exist 
anywhere - HSS User Unknown", and then forward to P-CSCF and the UE.

My question is why I-CSCF did not trigger a DNS/ENUM query after receiving LIA?

attached is the Kamailio configuration that we are using, please advise.

- RBK

On Wed, Jun 24, 2020 at 11:52 PM Valentin Christoph 
mailto:christoph.valen...@kontron.com>> wrote:

>

> I don't know your DNS server nor your addressing plan. Probably to

> modify / add some zone files

>

> You must be sure to forward the ranges of telephone numbers that you want to 
> forward to the other domain.

>

> Actually, it DID the request to the DNS/ENUM, but the result was "keep the 
> domain as it is" or "replace it by the own domain"

>

> If the DNS/ENUM query did NOT succeed, then you would run into route(PSTN) 
> and the request would be forwarded to "good old CS domain" to find a 
> destination.

>

> -Ursprüngliche Nachricht-

> Von: sr-users 
> mailto:sr-users-boun...@lists.kamailio.org>>
>  Im Auftrag von

> BALL SUN

> Gesendet: Mittwoch, 24. Juni 2020 16:49

> An: Kamailio (SER) - Users Mailing List 
> mailto:sr-users@lists.kamailio.org>>

> Betreff: Re: [SR-Users] Kamailio

Re: [SR-Users] Does kamailio sip domain makes sense if i defined some alias

2020-06-26 Thread David Villasmil

Id love to help, but I didn’t understand the problem:

- you had configured your kamailio with IP 1.2.3.4 and ALIAS sip.fqdn.com
- DNS on sip.fqdn.com returned 1.2.3.4
All good.

Then the DNS pointer changed 1.2.3.4 to 4.3.2.1.

Is that the question?
If so, I assume the IP where kamailio is running also changed the ip to
4.3.2.1.

You just need to change the listen IP. If you had an ALIAS to the IP as
well you also need to change that. The alias of the domain you don’t need
to change.

I hope I understood your question.

On Fri, 26 Jun 2020 at 17:24, PICCORO McKAY Lenz 
wrote:

> in kamctlrc i defined an DOMAIN but later that ip was updated..
>
> so my question: it make sense inclusivelly if already defined more
> allias in kamailiorc.cfg  file?
>
> in case of more complicated response,, anybody can explain me about that
> why?
>
> --
> Lenz McKAY Gerardo (PICCORO)
> http://qgqlochekone.blogspot.com
>
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Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [SR-Users] rtpengine inserted random line into sdp

2020-06-26 Thread Andrew Chen

Interesting.  Thanks for the tidbit.  I will play around with
msg_apply_changes and find the right place for it.

On Fri, Jun 26, 2020 at 9:01 AM Richard Fuchs  wrote:

> Rtpengine takes and replaces the entire SDP body, so if there's another
> module also manipulating the SDP, one module won't see the changes made by
> the other one unless you call msg_apply_changes in between, leaving you
> with bits of string in the wrong places.
>
> Cheers
>
> On 26/06/2020 08.50, Andrew Chen wrote:
>
> Thanks Richard.
>
> I do have the nathelper module running but not rtpproxy.  How does
> nathelper cause this issue?
>
> On Fri, Jun 26, 2020 at 8:27 AM Richard Fuchs  wrote:
>
>> Looks like you're using both rtpengine and some other SDP-modifying
>> module together, such as nathelper or rtpproxy, without calling
>> msg_apply_changes() in between.
>>
>> Cheers
>>
>> On 25/06/2020 16.46, Andrew Chen wrote:
>>
>> Hi forum,
>>
>> I'm starting my rtpengine project and I'm facing a strange problem with
>> rtpengine.  I am seeing this in the SDP part of the INVITE:
>>
>> a=rtcp:52021
>>
>> a=rtcp-mux
>>
>> 2001:470:7:3A7:0:0:0:2
>>
>> a=direction:active
>>
>> a=oldmediaip:54.153.25.234
>>
>> As you can see there is a random insert of the local kamailio IP in there
>> and I'm have tried the following to remove it but failed to succeed:
>>
>> - Remove in the offer the combination of ICE and
>> replace-session-connection flags
>> - Switched between rtpengine_offer and rtpengine_manage
>> - Restarted ngcp-rtpengine-daemon process
>> - Used sdpops function sdp_remove_line_by_preifx()
>>
>> Currently running Kamailio 5.1.2.
>>
>> I am running out of ideas.
>>
>> Any suggestions would be greatly appreciated.
>>
>> --
>> Andy Chen
>> Sr. Telephony Lead Engineer
>> achen@ fuze.com
>>
>>
>>
>> *Confidentiality Notice: The information contained in this e-mail and any
>> attachments may be confidential. If you are not an intended recipient, you
>> are hereby notified that any dissemination, distribution or copying of
>> this
>> e-mail is strictly prohibited. If you have received this e-mail in error,
>> please notify the sender and permanently delete the e-mail and any
>> attachments immediately. You should not retain, copy or use this e-mail or
>> any attachment for any purpose, nor disclose all or any part of the
>> contents to any other person. Thank you.*
>>
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>>
>
>
> --
> Andy Chen
> Sr. Telephony Lead Engineer
> 415 516 5535 (M)
> achen@ fuze.com
>
>
> *Confidentiality Notice: The information contained in this e-mail and any
> attachments may be confidential. If you are not an intended recipient, you
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>
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-- 
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
achen@ fuze.com

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Re: [SR-Users] rtpengine inserted random line into sdp

2020-06-26 Thread Andrew Chen

Thanks Richard.

I do have the nathelper module running but not rtpproxy.  How does
nathelper cause this issue?

On Fri, Jun 26, 2020 at 8:27 AM Richard Fuchs  wrote:

> Looks like you're using both rtpengine and some other SDP-modifying module
> together, such as nathelper or rtpproxy, without calling
> msg_apply_changes() in between.
>
> Cheers
>
> On 25/06/2020 16.46, Andrew Chen wrote:
>
> Hi forum,
>
> I'm starting my rtpengine project and I'm facing a strange problem with
> rtpengine.  I am seeing this in the SDP part of the INVITE:
>
> a=rtcp:52021
>
> a=rtcp-mux
>
> 2001:470:7:3A7:0:0:0:2
>
> a=direction:active
>
> a=oldmediaip:54.153.25.234
>
> As you can see there is a random insert of the local kamailio IP in there
> and I'm have tried the following to remove it but failed to succeed:
>
> - Remove in the offer the combination of ICE and
> replace-session-connection flags
> - Switched between rtpengine_offer and rtpengine_manage
> - Restarted ngcp-rtpengine-daemon process
> - Used sdpops function sdp_remove_line_by_preifx()
>
> Currently running Kamailio 5.1.2.
>
> I am running out of ideas.
>
> Any suggestions would be greatly appreciated.
>
> --
> Andy Chen
> Sr. Telephony Lead Engineer
> achen@ fuze.com
>
>
>
> *Confidentiality Notice: The information contained in this e-mail and any
> attachments may be confidential. If you are not an intended recipient, you
> are hereby notified that any dissemination, distribution or copying of this
> e-mail is strictly prohibited. If you have received this e-mail in error,
> please notify the sender and permanently delete the e-mail and any
> attachments immediately. You should not retain, copy or use this e-mail or
> any attachment for any purpose, nor disclose all or any part of the
> contents to any other person. Thank you.*
>
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-- 
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
achen@ fuze.com

-- 
*Confidentiality Notice: The information contained in this e-mail and any

attachments may be confidential. If you are not an intended recipient, you

are hereby notified that any dissemination, distribution or copying of this

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attachments immediately. You should not retain, copy or use this e-mail or

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[SR-Users] Does kamailio sip domain makes sense if i defined some alias

2020-06-26 Thread PICCORO McKAY Lenz

in kamctlrc i defined an DOMAIN but later that ip was updated..

so my question: it make sense inclusivelly if already defined more
allias in kamailiorc.cfg  file?

in case of more complicated response,, anybody can explain me about that why?

-- 
Lenz McKAY Gerardo (PICCORO)
http://qgqlochekone.blogspot.com

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Re: [SR-Users] RTP voice failover

2020-06-26 Thread Mathieu Bodjikian

Hello,



Unless I'm mistaken, there is no "out-of-the-box" solution.


A way to do it would be an active/sdandby  with keepalived + virtual IP + HA 
Redis setup.


With Kubernetes, it could be possible, but I'm not sure it's well suited for a 
wide port range (usually 1 UDP ports), but I'm interested if somebody has 
already experience with it.




Kind regards,
Mathieu Bodjikian


De : sr-users  de la part de Evgeniy 

Envoyé : vendredi 26 juin 2020 15:38:40
À : sr-users@lists.kamailio.org
Objet : [SR-Users] RTP voice failover

Is that possible to implement RTP voice fail-over?

In case RTP node goes down - the call should not  be interrupted.
This is kinda strange - but client want to have super redundant system.
For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
nodes (yes - network traffic will increase)
Or maybe is a way to recover UDP connections after a crash and replace
node state.
Maybe use a Kubernetes tools for that purpose ?
Here what i found in mailing-list -
https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html

Many thanks for any hints to this topic.




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[SR-Users] How to add parameter ;user=phone in RURI

2020-06-26 Thread Minh Phan

Hi List,

I'm having trouble adding the parameter user=phone in the RURI. I have
tried to use the function add_uri_param from siputils module but it doesn't
work (see below). Here is the code I use to add user=phone parameter to
FROM, TO and RURI

uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com
;user=phone");
uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone");
$rU = $var(normalized_number);
*add_uri_param*("user=phone");


Thanks in advance for any hints,
Best regards,
Minh Phan
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Re: [SR-Users] RTP voice failover

2020-06-26 Thread Alex Balashov

RTPEngine has Redis-based RTP flow state sharing you may wish to look into.

—
Sent from mobile, with due apologies for brevity and errors.

> On Jun 26, 2020, at 9:39 AM, Evgeniy  wrote:
> 
> Is that possible to implement RTP voice fail-over?
> 
> In case RTP node goes down - the call should not  be interrupted.
> This is kinda strange - but client want to have super redundant system.
> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes 
> (yes - network traffic will increase)
> Or maybe is a way to recover UDP connections after a crash and replace node 
> state.
> Maybe use a Kubernetes tools for that purpose ?
> Here what i found in mailing-list - 
> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
> 
> Many thanks for any hints to this topic.
> 
> 
> 
> 
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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED

2020-06-26 Thread Martin W Woscek

Hello,

Just wanted to update what fixed our issue.
The "contact_user" which is "asterisk" that we were using for the Kamailio 
context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 
16 server was not a valid user in the IMS HSS database (FHoSS implementation).

However not being an expert in the IMS, IMHO, the IMS should have kicked out 
this INVITE as soon as credentials were checked at the front door of the IMS, 
but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the 
correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 
604 HSS user unknown being sent thru the reverse direction back to Asterisk.

Anyway, its always good to end Friday on a high note.

Thanks,
_Martin


From: sr-users  On Behalf Of Martin W 
Woscek
Sent: Wednesday, June 24, 2020 1:32 PM
To: Kamailio (SER) - Users Mailing List ; 
mico...@gmail.com
Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK 
messages to Asterisk (to tag)

Just to clarify what we are seeing, here is a packet from the INVITE from the 
S-CSCF (6060) to the P-CSCF(4060).  I have eliminated the S-CSCF as this packet 
looks correct.
The P-CSCF is not handling the request URI correctly, and simply sends it back 
to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri 
updated by the S-CSCF with the local IP address of the UE (192.168.20.1).  The 
P-CSCF should send this INVITE to the UE 
(703222888@192.168.20.1)

No. Arrival Time SourceDestination   Source Port 
Destination Port SIP to tag SIP from tag Method Record-Route Userinfo 
Request-URI Media Port Protocol Length Request-URI Host Port Owner Username 
Media Type Media Format Origin-Host Type   slice_type SIP from address User 
Part SIP from address Host Part SIP to address User Part SIP to address Host 
Part Contact URI User Part Contact URI Host Part Request-URI User Part 
Request-URI Host Part Request-URI Host Port Info
   1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128
192.168.20.12860604060
d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt
sip:703222@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP  2263   
63871 -  audio,video ITU-T G.711 
PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 
  70 asterisk.irisims.org   703222  
 ims522.irisims.org   asterisk  192.168.20.131  
  703222192.168.20.1  63871 Request: 
INVITE sip:703222@192.168.20.1:63871;transport=udp |

Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) 
on interface any, id 0
Linux cooked capture
Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128
User Datagram Protocol, Src Port: 6060, Dst Port: 4060
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:703222@192.168.20.1:63871;transport=udp SIP/2.0
Method: INVITE
Request-URI: sip:703222@192.168.20.1:63871;transport=udp
Request-URI User Part: 703222
Request-URI Host Part: 192.168.20.1
Request-URI Host Port: 63871
[Resent Packet: False]
Message Header
Record-Route: 

Route: 
Via: SIP/2.0/UDP 
192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0
Via: SIP/2.0/UDP 
192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1
Via: SIP/2.0/UDP 
192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335
From: 
;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d
To: 
Contact: 
Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660
[Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660]
CSeq: 19314 INVITE
Route: 
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, 
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 68
User-Agent: Asterisk PBX 16.6.0
Content-Type: application/sdp
Content-Length:  1121
Message Body


From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Martin W Woscek
Sent: Wednesday, June 24, 2020 11:21 AM
To: mico...@gmail.com; Kamailio (SER) - Users Mailing 
List mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK 
messages to Asterisk (to tag)

Hi,
The issue is the P should be sending to the UE not back to the I like the 
failed flow:
Asterisk16-[phone]  --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604  HSS user 
unknown]  to Asterisk16[phone]

The successful call should look

Re: [SR-Users] Question about textops

2020-06-26 Thread Henning Westerholt

Hi Mojtaba,

not looked in the code – but it could be related to be able to check with this 
function for request and replies. Replies will have the method name in their 
CSEQ.

If you have a requirement that is not available right now in the module, it can 
be of course extended (e.g. by pull request).

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users  On Behalf Of Mojtaba
Sent: Friday, June 26, 2020 12:43 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about textops

Any idea?
On Wed, 24 Jun 2020, 14:26 Mojtaba, mailto:mes...@gmail.com>> 
wrote:
Hello,
Some confusing things while developing in Kamailio:
In is_method_f function in textops module, the code checks the HDR_CSEQ_F in 
msg also, and if the method name appears in CSEQ_F, The result is returned TRUE.
These question will  arise why in this function, the CSEQ field in message will 
check?
if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)

{

LM_ERR("cannot parse cseq header\n");

return -1; /* should it be 0 ?!?! */

}

if(m->s==0)

return (get_cseq(msg)->method_id>len)?1:-1;

else

return (get_cseq(msg)->method_id==METHOD_OTHER

&& get_cseq(msg)->method.len==m->len

&& (strncasecmp(get_cseq(msg)->method.s, m->s,
m->len)==0))?1:-1;

I think it would be nice to add new function like: is_cseq_method_f in module 
to avoid some problem understanding.

--
--Mojtaba Esfandiari.S
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Re: [SR-Users] RTP voice failover

2020-06-26 Thread Alex Balashov

I would agree about the marginal utility.

—
Sent from mobile, with due apologies for brevity and errors.

> On Jun 26, 2020, at 11:00 AM, Sergiu Pojoga  wrote:
> 
> 
> The most thorough presentation about the topic that I know of was 1&1's where 
> BGP is used. 
> The task of detecting such a failure within a couple of seconds so that 
> user's won't hang up intuitively, avoid the failse-positive failure 
> detection, makes the entire task pretty daunting and marginally useful, IMO. 
> 
> https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High-Quality-Telephony-Using-A-Fail-Safe-Media-Relay-Setup.pdf
>  
> 
>> On Fri, Jun 26, 2020 at 9:39 AM Evgeniy  wrote:
>> Is that possible to implement RTP voice fail-over?
>> 
>> In case RTP node goes down - the call should not  be interrupted.
>> This is kinda strange - but client want to have super redundant system.
>> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 
>> nodes (yes - network traffic will increase)
>> Or maybe is a way to recover UDP connections after a crash and replace 
>> node state.
>> Maybe use a Kubernetes tools for that purpose ?
>> Here what i found in mailing-list - 
>> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
>> 
>> Many thanks for any hints to this topic.
>> 
>> 
>> 
>> 
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Re: [SR-Users] RTP voice failover

2020-06-26 Thread Karsten Horsmann

Hi Evgeniy,

you can for example save rtpengine sessions informations in redis and do an
active / passive setup for media nodes.

Rtpengine is capable to fetch up the session on startup from redis if you
point it to the same redis dB

Just an idea.

Cheers
Karsten

Evgeniy  schrieb am Fr., 26. Juni 2020, 15:39:

> Is that possible to implement RTP voice fail-over?
>
> In case RTP node goes down - the call should not  be interrupted.
> This is kinda strange - but client want to have super redundant system.
> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
> nodes (yes - network traffic will increase)
> Or maybe is a way to recover UDP connections after a crash and replace
> node state.
> Maybe use a Kubernetes tools for that purpose ?
> Here what i found in mailing-list -
> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
>
> Many thanks for any hints to this topic.
>
>
>
>
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Re: [SR-Users] RTP voice failover

2020-06-26 Thread Sergiu Pojoga

The most thorough presentation about the topic that I know of was 1&1's
where BGP is used.
The task of detecting such a failure within a couple of seconds so that
user's won't hang up intuitively, avoid the failse-positive failure
detection, makes the entire task pretty daunting and marginally useful,
IMO.

https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High-Quality-Telephony-Using-A-Fail-Safe-Media-Relay-Setup.pdf


On Fri, Jun 26, 2020 at 9:39 AM Evgeniy  wrote:

> Is that possible to implement RTP voice fail-over?
>
> In case RTP node goes down - the call should not  be interrupted.
> This is kinda strange - but client want to have super redundant system.
> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
> nodes (yes - network traffic will increase)
> Or maybe is a way to recover UDP connections after a crash and replace
> node state.
> Maybe use a Kubernetes tools for that purpose ?
> Here what i found in mailing-list -
> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
>
> Many thanks for any hints to this topic.
>
>
>
>
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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[SR-Users] RTP voice failover

2020-06-26 Thread Evgeniy


Is that possible to implement RTP voice fail-over?

In case RTP node goes down - the call should not  be interrupted.
This is kinda strange - but client want to have super redundant system.
For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 
nodes (yes - network traffic will increase)
Or maybe is a way to recover UDP connections after a crash and replace 
node state.

Maybe use a Kubernetes tools for that purpose ?
Here what i found in mailing-list - 
https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html


Many thanks for any hints to this topic.




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Re: [SR-Users] rtpengine inserted random line into sdp

2020-06-26 Thread Richard Fuchs

Rtpengine takes and replaces the entire SDP body, so if there's another 
module also manipulating the SDP, one module won't see the changes made 
by the other one unless you call msg_apply_changes in between, leaving 
you with bits of string in the wrong places.


Cheers

On 26/06/2020 08.50, Andrew Chen wrote:

Thanks Richard.

I do have the nathelper module running but not rtpproxy. How does 
nathelper cause this issue?


On Fri, Jun 26, 2020 at 8:27 AM Richard Fuchs > wrote:


Looks like you're using both rtpengine and some other
SDP-modifying module together, such as nathelper or rtpproxy,
without calling msg_apply_changes() in between.

Cheers

On 25/06/2020 16.46, Andrew Chen wrote:

Hi forum,

I'm starting my rtpengine project and I'm facing a strange
problem with rtpengine.  I am seeing this in the SDP part of the
INVITE:

a=rtcp:52021

a=rtcp-mux

2001:470:7:3A7:0:0:0:2

a=direction:active

a=oldmediaip:54.153.25.234


As you can see there is a random insert of the local kamailio IP
in there and I'm have tried the following to remove it but failed
to succeed:

- Remove in the offer the combination of ICE and
replace-session-connection flags
- Switched between rtpengine_offer and rtpengine_manage
- Restarted ngcp-rtpengine-daemon process
- Used sdpops function sdp_remove_line_by_preifx()

Currently running Kamailio 5.1.2.

I am running out of ideas.

Any suggestions would be greatly appreciated.

-- 
Andy Chen

Sr. Telephony Lead Engineer
achen@ fuze.com 



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--
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
achen@ fuze.com 


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Re: [SR-Users] rtpengine inserted random line into sdp

2020-06-26 Thread Richard Fuchs

Looks like you're using both rtpengine and some other SDP-modifying 
module together, such as nathelper or rtpproxy, without calling 
msg_apply_changes() in between.


Cheers

On 25/06/2020 16.46, Andrew Chen wrote:

Hi forum,

I'm starting my rtpengine project and I'm facing a strange problem 
with rtpengine.  I am seeing this in the SDP part of the INVITE:


a=rtcp:52021

a=rtcp-mux

2001:470:7:3A7:0:0:0:2

a=direction:active

a=oldmediaip:54.153.25.234


As you can see there is a random insert of the local kamailio IP in 
there and I'm have tried the following to remove it but failed to succeed:


- Remove in the offer the combination of ICE and 
replace-session-connection flags

- Switched between rtpengine_offer and rtpengine_manage
- Restarted ngcp-rtpengine-daemon process
- Used sdpops function sdp_remove_line_by_preifx()

Currently running Kamailio 5.1.2.

I am running out of ideas.

Any suggestions would be greatly appreciated.

--
Andy Chen
Sr. Telephony Lead Engineer
achen@ fuze.com 



*Confidentiality Notice: The information contained in this e-mail and any
attachments may be confidential. If you are not an intended recipient, you
are hereby notified that any dissemination, distribution or copying of 
this

e-mail is strictly prohibited. If you have received this e-mail in error,
please notify the sender and permanently delete the e-mail and any
attachments immediately. You should not retain, copy or use this e-mail or
any attachment for any purpose, nor disclose all or any part of the
contents to any other person. Thank you.*

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Re: [SR-Users] Question about textops

2020-06-26 Thread Mojtaba

Any idea?

On Wed, 24 Jun 2020, 14:26 Mojtaba,  wrote:

> Hello,
> Some confusing things while developing in Kamailio:
> In is_method_f function in textops module, the code checks the HDR_CSEQ_F
> in msg also, and if the method name appears in CSEQ_F, The result is
> returned TRUE.
> These question will  arise why in this function, the CSEQ field in message
> will check?
> if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)
> {
> LM_ERR("cannot parse cseq header\n");
> return -1; /* should it be 0 ?!?! */
> }
> if(m->s==0)
> return (get_cseq(msg)->method_id>len)?1:-1;
> else
> return (get_cseq(msg)->method_id==METHOD_OTHER
> && get_cseq(msg)->method.len==m->len
> && (strncasecmp(get_cseq(msg)->method.s, m->s,
> m->len)==0))?1:-1;
>
> I think it would be nice to add new function like: is_cseq_method_f in
> module to avoid some problem understanding.
>
> --
> --Mojtaba Esfandiari.S
>
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[SR-Users] kamailio and rtpengine in docker

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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED

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Re: [SR-Users] RTP voice failover

Re: [SR-Users] RTP voice failover

Re: [SR-Users] RTP voice failover

[SR-Users] RTP voice failover

Re: [SR-Users] rtpengine inserted random line into sdp

Re: [SR-Users] rtpengine inserted random line into sdp

Re: [SR-Users] Question about textops

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