Hi Gbenga,
though you did not share your full script, one thing that catched my eyes
is that $avp(modifiedSDP) is never set (you have $avp(originalSDP)
though...)
ex:
$avp(originalSDP) = $rb;
...
set_body("$avp(modifiedSDP)","application/sdp");
I would check that 1st.
Other than that, as far as I know, rtpengine_manage() modifies the SDP by
itself, I don't think you would need a call to set_body().
Remove those lines and add something like:
if (sdp_content()) {
$avp(originalSDP) = $rb;
}
Also, you might want to use the direction parameters for RTPEngine, as it
will tell RTPEngine that the first leg uses interface X and the second leg
uses interface Y.
Something like:
rtpengine_manage("... direction=external direction=internal");
You might also want to check the condition in route[SET_RTP_REPLY] :
if(isflagset(FLB_FROM_CORE_TO_WEBRTC)) {
xlog("L_INFO", "SET_RTP_REPLY | FROM CORE TO WEBRTC");
rtpengine_manage("replace-origin replace-session-connection RTP/SAVPF
ICE=remove ...");
}
When sending a reply to WebRTC Client, use ICE=force instead of ICE=remove
to force RTPEngine to generate the correct ICE candidates/flags with the
external IP address.
Hope this helps.
Atenciosamente / Kind Regards / Cordialement / Un saludo,
*Sérgio Charrua*
*www.kahea.ai <http://www.kahea.ai> / www.voip.pt <http://www.voip.pt>*
*OpenTelecom* - Consulting for Telecoms, Lda
Tel.: +351 <callto:+351+91+104+12+66>91 631 11 44
Email : *[email protected] <[email protected]>*
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On Mon, Feb 2, 2026 at 3:59 PM Gbenga Daramola via sr-users <
[email protected]> wrote:
> Hello,
> I need assistance with my kamailio, RTPEngine and Freeswitch
> implementation with WEBRTC. I am using a cloud instance with a private IP
> and a 1:1 NAT Public IP. I have setup the internal and external RTPEngine
> config
> interface = internal/192.168.xx.xx;external/ 192.168.xx.xx!8X.XXX.XXX.XXX
> For my normal SIP to SIP implementation, my SIP signaling and RTP traffic
> goes fine and I get to have two way audio. but for Webrtc in the SDP reply
> to my public webrtc client, I get to see the private IP in the client
> response if I force ICE, thus causing no audio negotiation since the public
> IP can't reach the private IP.
> I have been troubleshooting this, I am beginning to think I am doing it
> wrong. Please has anyone implemented this or something similar. I need to
> make this work and I have limited time.
> Could I also get the best method for detecting WebRTC reply and response
> in order to set the right RTPmanage flags as this is also a major issue.
> Below is a snippet of my config.
>
> route[SET_DIRECTION_FLAG] {
>
> if (is_in_subnet("$si", "192.168.0.0/16")) {
>
> if ($proto == "WS" || $proto == "WSS" || $ru =~
> "transport=(ws|wss)") {
> xlog("L_INFO", "[DIR] CORE → WEBRTC | src=$si proto=$proto
> ru=$ru\n");
> setflag(FLB_FROM_CORE_TO_WEBRTC);
> return;
> }
>
> xlog("L_INFO", "[DIR] CORE → SIP | src=$si proto=$proto ru=$ru\n");
> setflag(FLB_FROM_CORE_TO_SIP);
> return;
> }
>
> else {
>
> # Public → WebRTC
> if ($proto == "WS" || $proto == "WSS") {
> xlog("L_INFO", "[DIR] PUBLIC → WEBRTC | src=$si proto=$proto
> ru=$ru\n");
> setflag(FLB_FROM_PUBLIC_FROM_WEBRTC);
> return;
> }
>
> # Public → SIP
> xlog("L_INFO", "[DIR] PUBLIC → SIP | src=$si proto=$proto
> ru=$ru\n");
> setflag(FLB_FROM_PUBLIC_FROM_SIP);
> return;
> }
> }
>
> route[SET_RTP_REQUEST] {
> if (!is_method("UPDATE|INVITE")) {
> return 0;
> }
>
> if (sdp_content()) {
> $avp(originalSDP) = $rb;
> if(isflagset(FLB_FROM_PUBLIC_FROM_WEBRTC)) {
> xlog("L_INFO", "SET_RTP_REQUEST | FROM PUBLIC FROM
> WEBRTC");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP ICE=remove direction=external
> direction=internal");
> }
> if(isflagset(FLB_FROM_PUBLIC_FROM_SIP)) {
> xlog("L_INFO", "SET_RTP_REQUEST | FROM PUBLIC FROM
> SIP");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP ICE=remove direction=external
> direction=internal");
> }
> if(isflagset(FLB_FROM_CORE_TO_WEBRTC)) {
> xlog("L_INFO", "SET_RTP_REQUEST | FROM CORE TO
> WEBRTC");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/SAVPF ICE=force direction=internal
> direction=external");
> }
> if(isflagset(FLB_FROM_CORE_TO_SIP)) {
> xlog("L_INFO", "SET_RTP_REQUEST | FROM CORE TO
> SIP");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP ICE=remove direction=internal
> direction=external");
> }
> set_body("$avp(modifiedSDP)","application/sdp");
> }
> }
>
> route[SET_RTP_REPLY] {
> xlog("L_INFO", "SET_RTP_REPLY | ENTERING THE ROUTE BLOCK");
>
> if (sdp_content()) {
> $avp(originalSDP) = $rb;
> if(isflagset(FLB_FROM_PUBLIC_FROM_WEBRTC)) {
> xlog("L_INFO", "SET_RTP_REPLY | FROM PUBLIC FROM
> WEBRTC");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP ICE=remove direction=external
> direction=internal");
> }
> if(isflagset(FLB_FROM_PUBLIC_FROM_SIP)) {
> xlog("L_INFO", "SET_RTP_REPLY | FROM PUBLIC FROM
> SIP");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP ICE=remove direction=external
> direction=internal");
> }
> if(isflagset(FLB_FROM_CORE_TO_WEBRTC)) {
> xlog("L_INFO", "SET_RTP_REPLY | FROM CORE TO
> WEBRTC");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/SAVPF ICE=remove direction=internal
> direction=external");
> }
> if(isflagset(FLB_FROM_CORE_TO_SIP)) {
> xlog("L_INFO", "SET_RTP_REPLY | FROM CORE TO SIP");
> rtpengine_manage("replace-origin
> replace-session-connection RTP/AVP direction=internal direction=external");
> }
> set_body("$avp(modifiedSDP)","application/sdp");
> }
>
> if ($rs=~"[3-6][0-9][0-9]") {
> rtpengine_manage();
> }
> }
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