[SR-Users] kamailio + asterisk + webrtc

2018-10-22 Thread arish haque
Hi all,
I am using kamailio as sip server and asterisk as media server behind
pf-sense
firewall (NAT) with a public ip.
Sip signalling as well as rtp packets are flowing correctly when the
endpoints are within
a network. But when trying from outside network sip signalling is working
perfectly
but there is no RTP.

Public ip behind pfsense -> 182.70.xx.yy
kamailio+rtpengine and asterisk servers are on private ips --> 192.168.1.x
kamailio ip - 192.168.1.x
asterisk ip - 192.168.1.y

Please, feel free to ask for more information.

Thanks & Regards,
Arish Haque
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Re: [SR-Users] kamailio + asterisk + webrtc

2018-10-23 Thread arish haque
Hi Fred,
I have advertised the udp port in kamailio.cfg like below -
listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060
But how to advertise rtp Ports??

I have attached sip.conf

*Asterisk 1\*

*   \ -- **Kamailio+RTPPROXY ** FW (pfsense
**182.75.xx.yy  public-IP)**--- INTERNET**\
**(private ip - 192.168.1.x)*

*Asterisk n--\  *



*Th**anks & Regards,*

*Arish Haque*


sip.conf
Description: Binary data
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[SR-Users] kamailio + asterisk + webrtc

2018-10-23 Thread arish haque
Hi,
Thank you for the reply!
I tried advertising udp port like this listen=udp:192.168.1.x:5060
advertise 182.70.xx.yy:5060.
How to advertise RTP ports?
Kamailio+RTPengine on a Centos s/m with a priv ip - 192.168.1.x
Asterisk is on another CentOs s/m with a priv ip - 192.168.1.y
pfsense on pub ip - 182.75.xx.yy
PORT FORWARDING on pfsense-
sip and rtp ports to kamailio+rtpengine server 192.168.1.x.
sip.conf


[general]
context=external
realm=192.168.1.y

bindaddr=0.0.0.0:5080
tcpenable=no
allowoverlap=no
transport=udp,wss,ws
srvlookup=yes
qualify=yes
directmedia=no
canreinvite=no
;-NAT SUPPORT-

externip=182.75.xx.yy
localnet=192.168.1.0/255.255.255.0
nat=yes
;nat=no   //tried but no RTP
;nat=force_rport,comedia  //tried but no RTP

;---RTP SUPPORT
rtpkeepalive=30
disallow=all
allow=gsm
allow=ulaw
allow=opus
allow=alaw
allow=g723
==
How to make RTPengine listen to public interface-- 182.75.181.70?

With Regards,
Arish Haque
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[SR-Users] kamailio+asterisk+webrtc

2018-10-23 Thread arish haque
Hi,
Thank you for the reply!
I tried advertising udp port like this listen=udp:192.168.1.x:5060
advertise 182.70.xx.yy:5060.
How to advertise RTP ports?
Kamailio+RTPengine on a Centos s/m with a priv ip - 192.168.1.x
Asterisk is on another CentOs s/m with a priv ip - 192.168.1.y
pfsense on pub ip - 182.75.xx.yy
PORT FORWARDING on pfsense-
sip and rtp ports to kamailio+rtpengine server 192.168.1.x.
How to make RTPengine listen on public interface-- 182.75.181.70?

sip.conf


[general]
context=external
realm=192.168.1.y

bindaddr=0.0.0.0:5080
tcpenable=no
allowoverlap=no
transport=udp,wss,ws
srvlookup=yes
qualify=yes
directmedia=no
canreinvite=no
;-NAT SUPPORT-

externip=182.75.xx.yy
localnet=192.168.1.0/255.255.255.0
nat=yes
;nat=no   //tried but no RTP
;nat=force_rport,comedia  //tried but no RTP

;---RTP SUPPORT
rtpkeepalive=30
disallow=all
allow=gsm
allow=ulaw
allow=opus
allow=alaw
allow=g723
==
and i am using asterisk realtime support for mysql to store peer details.

With Regards,
Arish Haque
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Re: [SR-Users] kamailio + asterisk + webrtc

2018-10-25 Thread arish haque
Hi,
I am using this command for starting RTPengine -
/usr/sbin/rtpengine --table=0 --interface=192.168.1.x\!182.75.xx.yy
--listen-ng=127.0.0.1:2 --pidfile=/var/run/rtpengine.pid --log-level=6
--log-facility=local1

in kamailio.cfg --> listen=MY_SIP_ADDR advertise 182.75.xx.yy:5060
But still there is no RTP from outside network.

Still not able to sort out what is the issue.If pfsense is blocking rtp
packets or any other.
but i have very config on pfsense.

Thanks & Regards,
Arish Haque
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[SR-Users] Kamailio TCP/TLS Connection getting failed

2019-01-30 Thread arish haque
Hi All,
I am running kamailio 5.2.x on CentOS 7. Call is happening but
i am getting the below error continuously in kamailio logs.

WARNING: websocket [ws_frame.c:228]: encode_and_send_ws_frame(): TCP/TLS
connection get failed
Jan 30 17:57:01 localhost /usr/local/sbin/kamailio[22097]: ERROR: websocket
[ws_frame.c:765]: ping_pong(): sending keepalive

Any clue of why this is happening?

Thanks,
Arish Haque
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[SR-Users] kamailio + asterisk + webrtc

2019-02-01 Thread arish haque
Hi Daniel,
Within a network above tech stack is working fine for VOIP calls.
But when i am trying from outside there is no sign of RTP.


Cmd for rtp -
/sbin/rtpengine -p /var/run/rtpengine.pid
--interface=192.168.1.249\!106.51.78.78 --listen-ng 127.0.0.1:6 -m
5 -M 55000 --log-level 6 --log-facility local1

106.51.78.78 - public ip
192.168.1.249 - priv ip (kamailio + RTPEngine)

Kamailio config when call is b/w WebRTC --> WebRTC
rtpengine_manage("trust-address replace-origin replace-session-connection
direction=internal direction=external");

Here is SDP, captured on the client sitting outside.
v=0
o=root 1133801452 1133801452 IN IP4 106.51.78.78
s=Asterisk PBX 16.0.0
c=IN IP4 106.51.78.78
t=0 0
m=audio 50196 UDP/TLS/RTP/SAVPF 0 8 111 9 126
a=maxptime:60
a=ice-ufrag:5e13a1292fc0b7163d49328b7516f763
a=ice-pwd:4fa5e003195492fa165020c81eff9346
a=candidate:Hc0a801ac 1 UDP 2130706431 192.168.1.172 46810 typ host
a=candidate:Hc0a801ac 2 UDP 2130706430 192.168.1.172 46811 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256
AA:5A:51:BD:4C:53:65:E4:2B:EC:EB:BF:A9:07:DD:60:E3:46:D8:26:6D:04:C8:21:8B:B9:81:37:3A:EB:55:C0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=sendrecv
a=rtcp:50196
a=rtcp-mux
a=ptime:20
a=candidate:T7Tqd3oKM4NvATH3 1 UDP 2097152255 106.51.78.78 50196 typ host

Looking forward for help from you guys.

Thanks & Regards,
Arish Haque
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[SR-Users] authentication using auth_db module

2019-04-18 Thread arish haque
Hi all,
I'm trying to establish mysql database connection inside request route
(at runtime) based on the type of user, as credentials are stored
in multiple databases (may be different servers) for the users.
Therefore, I am trying to find out that is there anyway to connect to mysql
dynamically in request route or pass 'db_url' parameter dynamically to
auth_db module.
Hope this explains the requirement.
Looking for suggestions.

With Regards,
Arish
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Re: [SR-Users] authentication using auth_db

2019-04-20 Thread arish haque
Hi David,
Thanks for the reply.
I checked out sqlops module, this will solve the problem
only when I get the authentication password (secret)
of the user inside request route. For this I tried checking
kamailio pseudovars but haven't found one. Therefore, I am
sticking with auth_db module.If this is possible by any other way
do let me know.

thanks,
Arish
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Re: [SR-Users] authentication using auth_db

2019-04-21 Thread arish haque
Hi David,
This can solve the issue but depth(records) of that view would be too much.
The architecture is like this:
One kamailio fronting multiple asterisk. Each of asterisk boxes have their
Specific users databases. Thus, I'm
trying to create db connections inside request route or completely write my
own authentication service if I get the user's
Password.
So, that would be great if I would be able to pass the db_url to auth_db
module in request  route.
Let me know what you think.

Thanks,
Arish
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[SR-Users] Sqlops

2019-08-07 Thread arish haque
Hi all,
I am querying mysql db inside routing block in kamailio.cfg

ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on
query: Unknown column '8d77' in 'where clause' (1054)
Aug  7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: 
[db_query.c:181]: db_do_raw_query(): error while submitting query
Aug  7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: sqlops
[sql_api.c:265]: sql_do_query(): cannot do the query [select * from
vca_asterisk where front_id=8d77]

I'm getting the above error message when, variable front_id=8d77 is
alphanumeric.
It's working fine when front_id is pure numeric. eg: front_id=877
Database column front_id is VARCHAR(255).

Any suggestions to solve this error?

Thanks,
Arish
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Re: [SR-Users] Sqlops

2019-08-08 Thread arish haque
Hi,
Passing the value within single quotes works.
Thanks for the help Fred.

With Regards,
Arish
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[SR-Users] No audio after hold and unhold event

2019-10-25 Thread arish haque
Hi all,
No Audio flows between the two parties when one side resumes the call after
putting on hold.
Configuration - WebRTC <--> Kamailio <--> Asterisk
Below is the warning I'm getting in asterisk console
WARNING[32388]: chan_sip.c:10425 process_sdp: Declining non-primary audio
stream: audio 10596 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112
113 101

Is this a codec issue or something else.

Thanks,
Arish
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