[SR-Users] WebRTC-UDP voicemail and call transfer

[Social Boh via sr-users]

Hello list,

I'm trying implementing a solution like:

https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg

The 2 problems I have a this moment are:

Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; 
the call go to voicemail but in the SDP still have WebRTC SDP


During a SIP UDP - WebRTC Call tranfer the call using phone button... 
the transfer don't work.


Any hint

Regards

--
---
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[SR-Users] Re: Kamailio, Asterisk and Parked calls

[Social Boh via sr-users]

Thank's all for your replies.

The problem is I can't find a pattern to do this. The Asterisk 
configuration is multitenant using chan_sip Channel. The customers 
parking the calls using the transfer sipphone button and pick up using 
BLF or calling the parking slot.


Each tenant have first 5 digits identifies him and then 3 digits more 
identifies each endpoint.


I tried using hash from URI and other configurations without success.

Regards

---
I'm SoCIaL, MayBe

El 25/04/2024 a las 3:44 p. m., Barry Flanagan escribió:

On Thu, 25 Apr 2024, at 21:17, Social Boh via sr-users wrote:

Hello list,

I'm using Kamailio to balance SIP request between two Asterisk. I have a
problem with Parking calls using Asterisk Application. When a user
parking a call, some times can't pick up parked call because the call to
pick up the Call go to the other Asterisk.

I'm using the 1 algorithm "hash over from URI" without success.


This is an Asterisk limitation/frustration really. Queues, Parking and Conferences need 
to run on the same server instance. So for a particular "tenant", for it to 
work you need to ensure that all parking for this tenant is directed to the same server. 
This also applies to any conference or queue.

If your individual endpoints can be recognised as being the same "tenant", try 
hashing over that tenant string and they will all get directed  to the same server.

If the above is not possible, but maybe the  number being dialled is 1:1 with 
the server (parking lot in your example) then you could hash over the $tU and 
achieve the same result.

In a nutshell, you need to find some common way that you can identify calls 
that should be grouped together, and then hash over that.

Hope this helps.

-Barry


Any hint?

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[SR-Users] Kamailio, Asterisk and Parked calls

[Social Boh via sr-users]

Hello list,

I'm using Kamailio to balance SIP request between two Asterisk. I have a 
problem with Parking calls using Asterisk Application. When a user 
parking a call, some times can't pick up parked call because the call to 
pick up the Call go to the other Asterisk.


I'm using the 1 algorithm "hash over from URI" without success.

Any hint?

--
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I'm SoCIaL, MayBe

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[SR-Users] Re: TLS module doesn't support TLSV1

[Social Boh via sr-users]

SIPTRACE module and SNGREP for TLS Capture:

https://www.voztovoice.org/?q=node/3020

Spanish

---
I'm SoCIaL, MayBe

El 4/04/2024 a las 3:12 a. m., Omar Atef via sr-users escribió:

Well, I've tried all versions lower than "TLSv1.2" and it didn't work just 
Kamailio running fine without activating TLS.

Also I wonder if there's a way to bug TLS SIP messages throw Kamailio itself because as 
you know it doesn't appear in "sngrep".

Thanks,
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[SR-Users] Re: Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?

[Social Boh via sr-users]

can you sto to spam on many mailing lists using excuse like this?

thank you

---
I'm SoCIaL, MayBe

El 28/03/2024 a las 1:24 a. m., Inextrix Technologies Pvt Ltd via 
sr-users escribió:

This question digs into the heart of Kamailio's functionality. It asks about:

Routing Logic: How Kamailio decides where to send SIP messages (calls, 
registrations, etc.) based on various factors like caller ID, destination 
number, or custom rules.
Customization: How you can modify this built-in logic to fit your specific 
needs. This could involve using scripting languages, manipulating SIP messages, 
or defining custom routing rules.
The discussion would be for developers who want to unlock Kamailio's full 
potential and tailor it for their unique VoIP deployments.
Visit us for More Info : https://inextrix.com/services/kamailio-development/
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[SR-Users] Re: RTP Engine with Kamailio

[Social Boh via sr-users]

Take a look at documentation module:

https://www.kamailio.org/docs/modules/5.8.x/modules/rtpengine.html#rtpengine.p.rtpengine_sock

you can define more RTPEngine Servers with or without weight for balancing

You can use rtpengine table too:

https://www.kamailio.org/docs/db-tables/kamailio-db-5.8.x.html#idm7211

I implemented Kamailio with more than 20 RTPENgine servers and works 
very well.


Regards

---
I'm SoCIaL, MayBe

El 22/03/2024 a las 1:16 p. m., Vishal Pai via sr-users escribió:

Hello Everyone.

I am looking for documentation to have multiple RTP Engine servers 
connected with kamailio using db url and balance the load accordingly.



Thanks

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[SR-Users] Re: best approach to setup SIP Session sharing between servers

[Social Boh via sr-users]

Better use corosync and pacemaker to keeep call audio flow during moving 
virtual IP using master-slave redis


---
I'm SoCIaL, MayBe

El 27/02/2024 a las 9:07 a. m., Sergio Charrua via sr-users escribió:

Thanks Alex & Olle!
I've done some tests with Keepalived for Virtual IP failover, and I 
found that it works more or less as you mentioned, Olle: if an Invite 
comes from Server #1, and that server gets offline, the BYE will 
indeed be correctly processed by Server #2.


Isn't DMQ supposed to handle this? as i understand, DMQ sends the SIP 
messages to other servers, so wouldn't it be enough? (I assume, I 
haven't tested DMQ yet )


*
*

*Sérgio Charrua*

*www.voip.pt *
Tel.: +351 91 631 11 44

Email : *sergio.char...@voip.pt*

This message and any files or documents attached are strictly 
confidential or otherwise legally protected.


It is intended only for the individual or entity named. If you are not 
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On Tue, Feb 27, 2024 at 2:20 PM Olle E. Johansson via sr-users 
 wrote:


In theory if you set the record-route to a DNS name and not an Ip
address, then endpoints should be able to failover within a dialog
- but as Alex says - transactions may fail. In many cases there
are retry timers to restart transactions that fail.

This means that if a call starts with an INVITE transaction on one
server, it should in theory be able to find another server for the
BYE. Whether this DNS srv based fail over is implemented in
end-points is up to testing to prove. If not, then virtual IP
failover is your best friend.

/O

> On 27 Feb 2024, at 13:33, Alex Balashov via sr-users
 wrote:
>
> That would require transaction replication, which Kamailio
doesn't have.
>
> Most messages can be processed statelessly, so this isn't a huge
obstacle. However, CANCELs won't work.
>
>> On 27 Feb 2024, at 06:49, Sergio Charrua via sr-users
 wrote:
>>
>> Hi all!
>>
>> I am having some difficulties/doubts on what would be the best
approach to let multiple Kamailio servers share their SIP sessions
between each other.
>> The goal is to have HA on Kamailio cluster such that if a call
is received and initiated on Kamailio #1, and during any moment of
the call (before or while established) if the server #1 goes down
or Kamailio stops working for any reason, the call can be
processed by Kamailio server #n available in the cluster.
>> I do not want to mess with Virtual IPs or DNS, that is not the
point, but instead need to get a proper way to share SIP sessions
between Kamailio servers such that any server could continue the
call without issue.
>>
>> I have checked the Dialog and DMQ module, but I am not sure if
that is the way to go
>>
>> Could any one share a solution for this?
>>
>> Greatly appreciated.
>>
>> Cheers,
>>
>>
>> Sérgio Charrua
>> www.voip.pt 
>>
>>
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>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
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[SR-Users] Re: Freeswitch and Kamailio Integration.

[Social Boh via sr-users]

Hello,

what type of integration do you need?

Only media services like voicemail or?

Regards

---
I'm SoCIaL, MayBe

El 21/02/2024 a las 6:26 a. m., jayapal--- via sr-users escribió:

Greetings,

I am new to Kamailio and seeking assistance. I have installed FreeSWITCH 
version 1.10.11 and Kamailio version 5.6.5. I have two inquiries:

I aim to configure FreeSWITCH with Kamailio. Could someone kindly guide me on 
achieving this integration?

I am curious about the storage location of Kamailio's log files. Could someone 
please provide information regarding this?

Thank you in advance for your help and guidance.
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[SR-Users] Re: PRACK RFC3262

[Social Boh via sr-users]

So Kamailio can't generate PRACK packets, only relay

Thank you

Regards

---
I'm SoCIaL, MayBe

El 9/02/2024 a las 3:54 a. m., Ihor Olkhovskyi via sr-users escribió:

Kamailio is a SIP proxy, it's not generating packets.
So,correct answer would be yes, Kamailio supports relay of the 
PRACK-related packets fully


Le jeu. 8 févr. 2024 à 22:16, Social Boh via sr-users 
 a écrit :


Does Kamailio support the SIP PRACK method? If yes, How?

Regards

-- 
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--
Best regards,
Ihor (Igor)

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[SR-Users] PRACK RFC3262

[Social Boh via sr-users]

Does Kamailio support the SIP PRACK method? If yes, How?

Regards

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[SR-Users] Re: LCR logic

[Social Boh via sr-users]

Thank you for your reply.

Regards

---
I'm SoCIaL, MayBe

El 26/01/2024 a las 6:21 p. m., Juha Heinanen escribió:

Social Boh via sr-users writes:


so I have 3 lcr_rule_target use the same
lcr_rule entry but 3 different Gateways and priority. So the order of
rules would be longest Request-URI (same) and the priority but the
result is:

added matched_gws[0]=[3, 5, 200, 2193562]
added matched_gws[1]=[1, 5, 2, 6888519]
added matched_gws[2]=[2, 5, 50, 2159388]

same prefix but first rule target with priority 200 and then 2 and then
50. I think the result would be priority 2,50 and 100.

¿Or not?

matched_gws are sorted after those debug messages.

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[SR-Users] LCR logic

[Social Boh via sr-users]

Hello,

LCR module logic is :

When the function /load_gws()/ is called, matching gateways (that are 
not currently designated as defunct) are ordered for forwarding purposes 
as follows:


1.

   according to longest Request-URI user part match

2.

   according to tuple's priority

3.

   according to tuple's randomized weight

or, if priority_ordering parameter is set to value 1, as follows:

1.

   according to tuple's priority

2.

   according to tuple's randomized weight

and Smaller priority value means higher priority (highest priority value 
being 0 and lowest being 255). so I have 3 lcr_rule_target use the same 
lcr_rule entry but 3 different Gateways and priority. So the order of 
rules would be longest Request-URI (same) and the priority but the 
result is:


added matched_gws[0]=[3, 5, 200, 2193562]
added matched_gws[1]=[1, 5, 2, 6888519]
added matched_gws[2]=[2, 5, 50, 2159388]

same prefix but first rule target with priority 200 and then 2 and then 
50. I think the result would be priority 2,50 and 100.


¿Or not?

---
I'm SoCIaL, MayBe
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[SR-Users] Change in LCR Debug

[Social Boh via sr-users]

Hello,

using DEBUG with LCR module to see how the module apply the rules, I see 
he use Gateway Index in the list. The Gateway Index could be different 
to Gateway ID and maybe can confuse the correct interpretation about the 
use of the rules.


Is it possible change the code:

matched_gws[gw_index].gw_index = t->gw_index;
    matched_gws[gw_index].rule_id = 
rule->rule_id;
    matched_gws[gw_index].prefix_len = 
pl->prefix_len;
    matched_gws[gw_index].priority = 
t->priority;
    matched_gws[gw_index].weight = 
t->weight * (kam_rand() >> 8);

    matched_gws[gw_index].duplicate = 0;
    LM_DBG("added matched_gws[%d]=[%u %u, 
%u, %u, %u]\n", gw_index,
    t->gw_index, 
pl->prefix_len, t->priority,

matched_gws[gw_index].weight);

to use gateway ID and not gateway Index?

Thank you

Regards

--
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[SR-Users] Re: webRTC Test Client?

[Social Boh via sr-users]

Hello,

I'm using this one:

https://github.com/InnovateAsterisk/Browser-Phone

Regards

---
I'm SoCIaL, MayBe

El 2/01/2024 a las 10:46 a. m., Andy Newlands via sr-users escribió:

Hi,

I have a Freeswitch installation, fronted by a Kamailio proxy

I would like to configure Kamailio to bridge SIP between webRTC 
clients and Freeswitch.


The first issue I hit is how to set up a test webRTC client to make 
calls into Kamailio/FS.


Can anyone point me to a simple webRTC client I can use to make 
test webRTC/SIP calls into my setup?  Thanks


BTW: I'm not JS experienced (I work with Python, C#, C/C++).

Kind regards,

Andy

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[SR-Users] Re: refer-to to new INVITE

[Social Boh via sr-users]

Hello,

I don't think dlg_bridge apply in this case.

I receive the REFER from remote PBX/SWITCH and I don't need to create a 
NEW call between two endpoints.


dlg_bridge, bridge a INVITE between two endpoints using REFER, Kamailio 
call first endpoint and when the endpoint answer send him a REFER with 
the callee data.


Right?

---
I'm SoCIaL, MayBe

El 25/11/2023 a las 12:09 p. m., David Villasmil via sr-users escribió:

https://sr-users.sip-router.narkive.com/1lg6Ov5U/problem-initiating-a-call-with-dlg-bridge


On Sat, 25 Nov 2023 at 06:39, Alex Balashov via sr-users 
 wrote:


Hi,

You may wish to explore this function, and its RPC correlate:

dialog Module

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>
kamailio.org

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>
favicon.ico

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>



<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>


-- Alex

—
Sent from mobile, apologies for brevity and errors.


    On Nov 22, 2023, at 7:17 PM, Social Boh via sr-users
 wrote:

Hello,


I'd like to know if there is a way, from Kamailio side, take the
refer-to header Kamailio receive in a REFER request and create a
new INVITE whit this data; then bridge the call between the
originating user and the new destination user.

Thank you

Regards

-- 
---

I'm SoCIaL, MayBe

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[SR-Users] Re: WEBRTC-to-SIP & Vice Versa

[Social Boh via sr-users]

Hello,

wich kind od issue do you have?

Regards

---
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El 25/11/2023 a las 12:17 a. m., Vishal Pai via sr-users escribió:

Thanks Yuriy.

Basically here we need to use the path, nathelper module and when we 
receive the invite for the call we need to route it accordingly to the 
destination address.


Note - Here I am facing an issue in call specially from Asterisk PBX 
(UDP) - Kamailio (WSS).


Any reference would be highly appreciated.

On Thu, Nov 2, 2023 at 3:39 AM Yuriy G  wrote:

Most common rules are: Use Path
https://datatracker.ietf.org/doc/html/rfc3327
As well as use add_contact_alias/handle_ruri_alias from the
nathelper module or same pattern ( like add_path_receoved ) to not
destroy original Contact information, and, at the same time to
provide destination address where to send INVITE to...

On Wed, 1 Nov 2023, 20:28 Vishal Pai via sr-users,
 wrote:

Hello All

Good Day

Here I am trying to achieve something like this

WSS to UDP Ext registartion which works well using
https://github.com/havfo/WEBRTC-to-SIP/tree/master/etc
Note here i am forwarding the registartion to asterisk pbx
server and define WSS of kamailio

Now for calls

- Webrtc - SIP calls works

- SIP - Webrtc calls not working
Here I am getting the request of the call on kamailio but
kamailio send 404 not found to Asterisk PBX

Webrtc - Webrtc calls not working

Can anyone guide me on what I am doing wrong?

Thanks

Vishal P
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[SR-Users] Re: Online devel meeting on matrix - 14:30 UTC, Dec 5, 2023

[Social Boh via sr-users]

Maybe a topic would be add a module to process REFER SIP method at 
Kamailio Level like B2B opensips modules.


I have more than one customer Ask me to implement this type of solution..

Regards

---
I'm SoCIaL, MayBe

El 24/11/2023 a las 5:27 a. m., Daniel-Constantin Mierla via sr-users 
escribió:

Hello,

we should consider an online devel meeting sometime soon to summarize
what was done at (and still needs to be done after) devel meeting in
Dusseldorf and plan a bit the targets for next major release (5.8 or 6.0)?

If considered useful, I propose Dec 5 at 14:30UTC (15:30
Berlin/Paris/Madrid/Rome), but we can also look for other dates as well.

Topics to be discussed can be added at:

   -
https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings/2023a.md

Pull requests can be made by users without git access.

Cheers,
Daniel


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[SR-Users] refer-to to new INVITE

[Social Boh via sr-users]

Hello,

I'd like to know if there is a way, from Kamailio side, take the 
refer-to header Kamailio receive in a REFER request and create a new 
INVITE whit this data; then bridge the call between the originating user 
and the new destination user.


Thank you

Regards

--
---
I'm SoCIaL, MayBe

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[SR-Users] Re: sips to sip

[Social Boh via sr-users]

Hello DAvid,

can you attach a pcap file about one example call?

Regards

---
I'm SoCIaL, MayBe

El 20/11/2023 a las 1:44 p. m., David Villasmil via sr-users escribió:

Hello guys,

I have this setup where one side is TLS and the other UDP. Normally 
this works fine, but we have this provider sending sips as the schema 
everywhere (from, to, rr, contacts), kamailio sends the same sips to 
the upstream usp freeswitch.


My problem is when FS sends back a 200OK and kamailio forwards it back 
to the provider, the provider sends an ACK and kamailio can't match it 
with the dialog and doesn't know where to forward it.


i think this is happening because FS when is sees SIPS is setting the 
contact port as 5081 instead of the usual 5080...



Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


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[SR-Users] Re: Obsoleting unmaintained modules

[Social Boh via sr-users]

I use uri_db module and the does_uri_exist() function to know if a user 
exist or not.


I don't know if there is other function I can use with the same task

Regards

---
I'm SoCIaL, MayBe

El 14/11/2023 a las 7:30 a. m., Daniel-Constantin Mierla via sr-users 
escribió:

Hello,

during the Kamailio Developers Meeting 2023 in Dusseldorf that took
place last week, it was proposed to obsolete modules that seem to be
unmaintained and no activity about them was noticed during the past
years. It is quite some overhead in packaging them and trying to keep
them compiling when they have external dependencies, therefore such step
should spare some resources in the future.

The list (see below) was built based on the options of those present at
the meeting, now we want to discuss it on the larger communities of
developers and users. If you are using any of these modules or you think
any of them worth keeping, reply with the names of the modules that you
want to be kept.

The proposed action is to relocate the obsoleted modules to a new git
repository "kamailio-obsolete" to still keep some visibility to them and
in the eventually of future interest on any of them, it can be
reintroduced in the main repository.

  Next is the initial list of modules proposed to be considered obsolete:

- app_java
- app_lua_sr
- app_mono
- app_python
- app_sqlang
- auth_identity
- call_control
- db2_ldap
- db2_ops
- db_cassandra
- db_perldvdb
- dnssec
- domainpolicy
- h350
- mediaproxy
- osp
- peering
- print
- print_lib
- pua_xmpp
- ratelimit
- uid_auth_db
- uid_avp_db
- uid_domain
- uid_gflags
- uid_uri_db
- uri_db
- xmpp
- xprint

Cheers,
Daniel


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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

hello,

i only a example... not for production.

Regards

---
I'm SoCIaL, MayBe

El 16/10/2023 a las 10:37 a. m., Mirko Brankovic via sr-users escribió:
I'm maybe out of context here but i don't think you can remove ssrc 
headers from sdp since they are crucial in identification of the rtp 
streams


On Mon, Oct 16, 2023, 14:41 Social Boh via sr-users 
 wrote:


Hello,

in the rtpengine LOG i can see:

Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG:
[28vs0pbtdtu7udefti8m]: ... "*: [ "sdp-attr-remove-audio-ssrc::"
], "replace":* [ "origin", "session-connection" ],
"transport-protocol": "RTP/AVP", "rtcp-mux": [ "demux" ], "SDES":
[ "off" ], "call-id": "28vs0pbtdtu7udefti8m", "via-branch":
"z9hG4bK4978560", "received-from": [ "IP4", "186.98.231.105" ],
"from-tag": "h8l2da68sp", "command": "offer" }

but the ssrc lines still presents in the INVITE sent to sip device.

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió:

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + "
rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP
sdp-attr-remove-audio-ssrc:";

this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users
escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side?
You could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations
options, so you can use rtpengine to strip attributes that it
itself doesn't understand or process.

Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

in the rtpengine LOG i can see:

Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG: [28vs0pbtdtu7udefti8m]: 
... "*: [ "sdp-attr-remove-audio-ssrc::" ], "replace":* [ "origin", 
"session-connection" ], "transport-protocol": "RTP/AVP", "rtcp-mux": [ 
"demux" ], "SDES": [ "off" ], "call-id": "28vs0pbtdtu7udefti8m", 
"via-branch": "z9hG4bK4978560", "received-from": [ "IP4", 
"186.98.231.105" ], "from-tag": "h8l2da68sp", "command": "offer" }


but the ssrc lines still presents in the INVITE sent to sip device.

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió:

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux 
DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:";


this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You 
could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations options, 
so you can use rtpengine to strip attributes that it itself doesn't 
understand or process.


Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux 
DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:";


this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You 
could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations options, 
so you can use rtpengine to strip attributes that it itself doesn't 
understand or process.


Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Thank you to answer.

this is my SDP annex (no ice o related stuff):


Content-Length: 1031


/v=0/

/o=- 326299663336889577 2 IN IP4 XXX.XXX.XXX.XXX/

/s=-/

/t=0 0/

/a=extmap-allow-mixed/

/a=msid-semantic: WMS 078639f4-3b58-4a29-90ef-62d05ccf01a2/

/m=audio 21766 RTP/AVP 111 63 9 0 8 13 110 126/

/c=IN IP4 //XXX.XXX.XXX.XXX/

/a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level/

/a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time/

/a=extmap:3 
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/


/a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid/

/a=msid:078639f4-3b58-4a29-90ef-62d05ccf01a2 
4cacf939-a056-4a1a-a0c5-748aaf5e3d43/


/a=ssrc:805220451 cname:wrpkmkhX7zbjXfUu/

/a=ssrc:805220451 msid:078639f4-3b58-4a29-90ef-62d05ccf01a2 
4cacf939-a056-4a1a-a0c5-748aaf5e3d43/


/a=mid:0/

/a=rtpmap:111 opus/48000/2/

/a=fmtp:111 useinbandfec=1; minptime=10/

/a=rtcp-fb:111 transport-cc/

/a=rtpmap:63 red/48000/2/

/a=fmtp:63 111/111/

/a=rtpmap:9 G722/8000/

/a=rtpmap:0 PCMU/8000/

/a=rtpmap:8 PCMA/8000/

/a=rtpmap:13 CN/8000/

/a=rtpmap:110 telephone-event/48000/

/a=rtpmap:126 telephone-event/8000/

/a=sendrecv/

/a=rtcp:21767/

I think I can eliminate  a=extmap lines and maybe other lines but i 
don't really know where I have to put *sdp_remove_line_by_prefix* 
function//in the script/

/

Regards/
/

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 1:46 a. m., Karsten Horsmann escribió:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You could 
strip that of with rtpengine options.


Social Boh via sr-users  schrieb am Sa., 
14. Okt. 2023, 01:00:


Hello,

I'm trying to solve a problem with calls WebRTC/UDP. I think is a
fragmentation problem because udp/webRTC calls works without
problem. The INVITE from Kamailio to the UDP device not receive
any type of answer.

I'm trying to reduce the SDP annex size using:


  |sdp_remove_line_by_prefix|

for some lines unsuccessfully. Maybe I don't know where put this
function in the script.

Or, is there another way to solve this issue?

Thank you in advance

Regards


-- 
---

I'm SoCIaL, MayBe

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[SR-Users] UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

I'm trying to solve a problem with calls WebRTC/UDP. I think is a 
fragmentation problem because udp/webRTC calls works without problem. 
The INVITE from Kamailio to the UDP device not receive any type of answer.


I'm trying to reduce the SDP annex size using:


 |sdp_remove_line_by_prefix|

for some lines unsuccessfully. Maybe I don't know where put this 
function in the script.


Or, is there another way to solve this issue?

Thank you in advance

Regards


--
---
I'm SoCIaL, MayBe
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