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.orghttps://kamailio.org/docs/modules/devel/modules/rtpengine.html 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@lists.kamailio.org on behalf of Prashant Gupta prashant@farmguide.in Sent: Wednesday, January 9, 2019 10:36 AM To: sr-users@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:127.0.0.1:45038http://127.0.0.1:45038/") this in my kamailio cfg but I don;t know which port to use here. Any suggestions?

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@lists.kamailio.org on behalf of Serge S.Yuriev me@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@farmguide.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:127.0.0.1:45038http://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.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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@gmail.com phone: +34669448337

ᐧ

On Thu, Jan 10, 2019 at 12:33 PM Serge S. Yuriev me@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@lists.kamailio.org on behalf of Serge S.Yuriev me@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@farmguide.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

-- Serge S. Yuriev Senior VoIP engineer

---

Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users