sip.js and sdp error 488 about webrtc-to-sip

[havfo]

Hi, this setup is not built for bridging between WebRTC clients by default. The Kamailio config needs some modifications if that is to work. What call scenarios do you plan to support?

To disable WITH_ALWAYS_BRIDGE, change the line:
#!define WITH_ALWAYS_BRIDGE
to
##!define WITH_ALWAYS_BRIDGE

[balabanferhat]

My call scenarios are webrtc<>sip , webrtc<>webrtc, sip<>sip

If I change the line like you said, I get compile errors and cannot start kamailio

loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/
 0(7298) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression...
 0(7298) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line **619**, column 1: syntax error
 0(7298) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line **619**, column 1: invalid  route  statement
 0(7298) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line **622**, column 1-12:
ERROR: bad config file (3 errors)

These lines ,619 and 622 can be seen below

My config file is same as yours. Nothing changed only XXXX parts are changed.

route[RTP_BRIDGE] {
	#!ifdef WITH_ALWAYS_BRIDGE
	if (is_method("INVITE")) {
		if ($ru =~ "transport=ws") {
			xlog("L_INFO", "SIP -> WebRTC, bridging RTP->SRTP and adding ICE");
			rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-accept rtcp-mux-offer ICE=force UDP/TLS/RTP/SAVPF");
			t_on_reply("REPLY_FROM_WS");
		} else if ($proto =~ "ws") {
			xlog("L_INFO", "WebRTC -> SIP, bridging SRTP->RTP and removing ICE");
			rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-demux ICE=remove RTP/AVP");
			t_on_reply("REPLY_TO_WS");
		}
	}
	#!endif
}  <------------------619

# manage outgoing branches
branch_route[MANAGE_BRANCH] {   <------------------622

By the way, I am now trying only webrtc<>webrtc and I cannot get rid of this 488 error.

[havfo]

Ah, nice catch. A route block always needs to contain something, so put a logging statement or something there. Example:

route[RTP_BRIDGE] {
	xlog("L_INFO", "ROUTE: RTP_BRIDGE");
	#!ifdef WITH_ALWAYS_BRIDGE
...

Like I said, this configuration will not handle all those scenarios by default. The intended use of this configuration is to make WebRTC be able to call SIP and vice versa.

From your description of network issues, I suspect you haven't set up a turn server? The WebRTC clients shouldn't need a RTPEngine between them, the TURN server should be enough.

[balabanferhat]

Hi, thanks for your help. I managed to make calls webrtc<>webrtc, webrtc > sip and sip<>sip
but I can do it by changing configurations. How can I detect if call is webrtc<>webrtc and change configuration dynamically?

[havfo]

Hi, I just pushed out a new config you can try. It's called kamailio-bridging.cfg. The new config should be smarter in handling the different types of bridging needed based on your needs.

[balabanferhat]

Thks, it is much smarter now, and all settings work except video calls between sip<>ws and sip<>sip.

I tested phone calls they are all ok.

for ws<> ws I commented rtpengine_manage and succeeded like this setup. I think we dont need bridging for this?

route[SETUP_BRIDGING] {
	if(!has_totag()) {
		if ($proto =~ "ws") { # Coming from websocket
			if ($ru =~ "transport=ws") { # WebRTC > WebRTC
				xlog("L_INFO", "WebRTC > WebRTC");
				# rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force");
				# t_on_reply("REPLY_WS_TO_WS");
			} else { # WebRTC > SIP
				xlog("L_INFO", "WebRTC > SIP");
				rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-demux ICE=remove RTP/AVP");
				t_on_reply("REPLY_WS_TO_SIP");
			}
		} else { # Coming from SIP
			if ($ru =~ "transport=ws") { # SIP > WebRTC
				xlog("L_INFO", "SIP > WebRTC");
				rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-accept rtcp-mux-offer ICE=force RTP/SAVPF");
				t_on_reply("REPLY_SIP_TO_WS");
			} else { # SIP > SIP
				xlog("L_INFO", "SIP > SIP");
				rtpengine_manage("trust-address replace-origin replace-session-connection");
				t_on_reply("REPLY_SIP_TO_SIP");
			}
		}
	}
}

For sip clients I use linphone. And video is not working in ios and android clients. Has kamailo a configuration to enable video-media streams ? . I have seen that Zopier client has a setting rport that enables media and signaling. If I enable it video starts working. But in linphone there is no such a setting and Nat cannot be handled. Therefore, video works only if clients are in same network.

[havfo]

For your first question, you are right that you don't need bridging for all the cases, ws <> ws should be best without RtpEngine.

Regarding the video issue, I am not sure why this doesn't work. There is nothing in Kamailio that has antything to do with video in that sense. RtpEngine modifies the SDP, but doesn't disable streams. Kamailio by itself doesn't care what is in the SDP.

I just pushed out a config that should fix handling of NAT for SIP clients (haven't had time to test it).