9 Jan
2019
9 Jan
'19
8:36 a.m.
Hi,
I have the following architecture - SIP provider <-> Kamailio <-> Asterisk
servers
Currently I have everything setup and incoming calls from Sip are routed to
my asterisk server. The issue is however that when I answer the call, there
is no media in the call. I have tried connecting with a normal local
extension(not SIP,eg 1001) and there is a normal flow of media.
When i try to sniff my connection via Wireshark on the asterisk server,
there is an outflow of RTP packets but the same RTP traffic does not appear
on the Wireshark of my Kamailio server connection.
I am not sure if this is an RTP engine issue and how to resolve this.
I have -
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:45038")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
9 Jan
9 Jan
8:48 a.m.
Hello,
1- Probably, there isnt any command to active rtpengine in your conf. check your
configuration that do you active rtpengine_manage/offer/answer() on INVITE
2- Rtpengine service is working?
3- Kamailio can connect to rtpengine service , you can see systemlog/message
4- take a look your pcap , rtpengine can change ip address in sdp
5 - do you set "replace-origin replace-session-connection" parameters to
rtpengine?
rtpengine doc : https://kamailio.org/docs/modules/devel/modules/rtpengine.html
rtpengine Module -
kamailio.org<https://kamailio.org/docs/modules/devel/modules/rtpengine.h…
This is a module that enables media streams to be proxied via an RTP proxy. The only RTP
proxy currently known to work with this module is the Sipwise rtpengine https ...
kamailio.org
Good luck
Yasin CANER
________________________________
From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Prashant Gupta
<prashant(a)farmguide.in>
Sent: Wednesday, January 9, 2019 10:36 AM
To: sr-users(a)lists.kamailio.org
Subject: [SR-Users] No Media in SIP Incoming calls
Hi,
I have the following architecture - SIP provider <-> Kamailio <-> Asterisk
servers
Currently I have everything setup and incoming calls from Sip are routed to my asterisk
server. The issue is however that when I answer the call, there is no media in the call. I
have tried connecting with a normal local extension(not SIP,eg 1001) and there is a normal
flow of media.
When i try to sniff my connection via Wireshark on the asterisk server, there is an
outflow of RTP packets but the same RTP traffic does not appear on the Wireshark of my
Kamailio server connection.
I am not sure if this is an RTP engine issue and how to resolve this.
I have -
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:45038<http://127.0.0.1:45038/>")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
10:11 a.m.
Hello,
Kamailio support 2 backend for rtp. RTP Engine and RTP proxy. it depends your
configuration. I read from your post that
"I am not sure if this is an RTP engine issue and how to resolve this." . I
thought it is rtpengine.
Indeed , they are different services and has different configuration that do same thing
that relays rtp packets. you can find examples from documentation.
I didnt get internal interface what is mean? but if rtpproxy/engine relays to Asterisk on
-lo interface , you should listen your -lo interface by Wireshark.
Good luck.
Yasin CANER
________________________________
From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Serge S.Yuriev
<me(a)nevian.org>
Sent: Thursday, January 10, 2019 11:30 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] No Media in SIP Incoming calls
Hi
- I see you mentioned rtpengine but config shows rtpproxy - which one you use? Maybe you
mixed things?
- Have you tried to capture internal interface of kamailio machine? Is RTP there?
--
Wbr, Serge via mobile
09.01.2019, 10:37, "Prashant Gupta" <prashant(a)farmguide.in>in>:
Hi,
I have the following architecture - SIP provider <-> Kamailio <-> Asterisk
servers
Currently I have everything setup and incoming calls from Sip are routed to my asterisk
server. The issue is however that when I answer the call, there is no media in the call. I
have tried connecting with a normal local extension(not SIP,eg 1001) and there is a normal
flow of media.
When i try to sniff my connection via Wireshark on the asterisk server, there is an
outflow of RTP packets but the same RTP traffic does not appear on the Wireshark of my
Kamailio server connection.
I am not sure if this is an RTP engine issue and how to resolve this.
I have -
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:45038<http://127.0.0.1:45038/>")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
1:31 p.m.
Hello,
I mean Internal facing to Asterisk(s)
External - facing to Provider(s)
On 10/01/2019 12:11, YASIN CANER wrote:
Hello,
Kamailio support 2 backend for rtp. RTP Engine and RTP proxy. it depends
your configuration. I read from your post that
"I am not sure if this is an _RTP engine _issue and how to resolve
this." . I thought it is rtpengine.
Indeed , they are different services and has different configuration
that do same thing that relays rtp packets. you can find examples from
documentation.
I didnt get internal interface what is mean? but if rtpproxy/engine
relays to Asterisk on -lo interface , you should listen your -lo
interface by Wireshark.
Good luck.
Yasin CANER
------------------------------------------------------------------------
*From:* sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of
Serge S.Yuriev <me(a)nevian.org>
*Sent:* Thursday, January 10, 2019 11:30 AM
*To:* Kamailio (SER) - Users Mailing List
*Subject:* Re: [SR-Users] No Media in SIP Incoming calls
Hi
- I see you mentioned rtpengine but config shows rtpproxy - which one
you use? Maybe you mixed things?
- Have you tried to capture internal interface of kamailio machine? Is
RTP there?
--
Wbr, Serge via mobile
09.01.2019, 10:37, "Prashant Gupta" <prashant(a)farmguide.in>in>:
> Hi,
> I have the following architecture - SIP provider <-> Kamailio <->
> Asterisk servers
> Currently I have everything setup and incoming calls from Sip are
> routed to my asterisk server. The issue is however that when I answer
> the call, there is no media in the call. I have tried connecting with
> a normal local extension(not SIP,eg 1001) and there is a normal flow
> of media.
> When i try to sniff my connection via Wireshark on the asterisk
> server, there is an outflow of RTP packets but the same RTP traffic
> does not appear on the Wireshark of my Kamailio server connection.
> I am not sure if this is an RTP engine issue and how to resolve this.
> I have -
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038
> <http://127.0.0.1:45038/>")
> this in my kamailio cfg but I don;t know which port to use here.
> Any suggestions?
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
--
Serge S. Yuriev
Senior VoIP engineer
3:10 p.m.
Hello,
As previously stated, you probably are not activating you rtpproxy. When
kamailio starts up, it checks whether it can connect to the local rtpproxy.
Check your kamailio log for errors.
You also need to have rtpproxy running on your kamailio box, on the
specified port (or setting the correct port on the config file).
If your asterisk is on a private network, you will need to start rtpproxy
specifying the outside and the inside (private) rtp ips, like:
rtpproxy -l [PUBLIC-IP] [PRIVATE-IP] -s udp:127.0.0.1:7722 -u rtpproxy
rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -d DBUG LOG_LOCAL0
and in your config, (my case) something like:
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")
then, when calls are being established, you need to check where it's
coming from and where it's going, and execute rtproxy with the parameter of
what IP to use, something like:
public ip:
rtpproxy_manage("cor", "[PUBLIC-IP]");
private ip:
rtpproxy_manage("co", "[PRIVATE-IP]");
good luck.
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
On Thu, Jan 10, 2019 at 12:33 PM Serge S. Yuriev <me(a)nevian.org> wrote:
Hello,
I mean Internal facing to Asterisk(s)
External - facing to Provider(s)
On 10/01/2019 12:11, YASIN CANER wrote:
Hello,
Kamailio support 2 backend for rtp. RTP Engine and RTP proxy. it depends
your configuration. I read from your post that
"I am not sure if this is an _RTP engine _issue and how to resolve
this." . I thought it is rtpengine.
Indeed , they are different services and has different configuration
that do same thing that relays rtp packets. you can find examples from
documentation.
I didnt get internal interface what is mean? but if rtpproxy/engine
relays to Asterisk on -lo interface , you should listen your -lo
interface by Wireshark.
Good luck.
Yasin CANER
------------------------------------------------------------------------
*From:* sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of
Serge S.Yuriev <me(a)nevian.org>
*Sent:* Thursday, January 10, 2019 11:30 AM
*To:* Kamailio (SER) - Users Mailing List
*Subject:* Re: [SR-Users] No Media in SIP Incoming calls
Hi
- I see you mentioned rtpengine but config shows rtpproxy - which one
you use? Maybe you mixed things?
- Have you tried to capture internal interface of kamailio machine? Is
RTP there?
--
Wbr, Serge via mobile
09.01.2019, 10:37, "Prashant Gupta" <prashant(a)farmguide.in>in>:
> Hi,
> I have the following architecture - SIP provider <-> Kamailio <->
> Asterisk servers
> Currently I have everything setup and incoming calls from Sip are
> routed to my asterisk server. The issue is however that when I answer
> the call, there is no media in the call. I have tried connecting with
> a normal local extension(not SIP,eg 1001) and there is a normal flow
> of media.
> When i try to sniff my connection via Wireshark on the asterisk
> server, there is an outflow of RTP packets but the same RTP traffic
> does not appear on the Wireshark of my Kamailio server connection.
> I am not sure if this is an RTP engine issue and how to resolve this.
> I have -
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038
> <http://127.0.0.1:45038/>")
> this in my kamailio cfg but I don;t know which port to use here.
> Any suggestions?
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
--
Serge S. Yuriev
Senior VoIP engineer
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
1937
days inactive
1938
days old
5 comments
5 participants
participants (5)
-
David Villasmil -
Prashant Gupta -
Serge S. Yuriev -
YAS0 CANER -
YASIN CANER