[SR-Users] kamailio with mediaproxy-ng, 488 Not Acceptable Here
Richard Fuchs
rfuchs at sipwise.com
Thu Feb 6 14:43:10 CET 2014
Hey,
What mediaproxy-ng can do (which other RTP proxies don't do) is
translate RTP/AVP (from regular SIP endpoints) to and from RTP/SAVPF
(encrypted RTP from WebRTC), which is what people usually use it for.
Assuming that ICE does its job correctly, two WebRTC endpoints should be
able to communicate directly with each other (without RTP proxy), even
if a firewall is involved. But I understand that in some cases even ICE
might fail.
There's two things I see wrong with the resulting SDP body, both related
to the last stable version of MP-NG not supporting BUNDLE. If you could
try adding an SDP rewrite rule to your kamailio config to remove the
a=group:... line. If that doesn't work, also try disabling video when
making the call.
Alternatively, you can try building MP-NG from this branch:
https://github.com/sipwise/mediaproxy-ng/tree/rfuchs/3.0
This is currently under heavy development, but it should support BUNDLE
just enough to make this work.
cheers
On 02/05/14 11:23, Mihai Marin wrote:
> Hello,
> I'm trying the simplest case first. I don't understand why you are
> saying that most of the people don't use mediaproxy-ng
> for WebRTC to WebRTC calls. If my firewall is a restrictive one I need
> to use turn server and mediaproxy-ng can do turn too? Probably I'm not
> seeing the big picture.
>
> Regarding the problem with Incompatible SDP I have attached the SDP
> before mp-ng and after:
>
> BEFORE mediaproxy-ng magic:
> 14(21473) DEBUG: websocket [ws_frame.c:650]: ws_frame_receive(): Rx SIP
> message:
> INVITE sip:bob at 93.187.138.214 <mailto:sip%3Abob at 93.187.138.214> SIP/2.0
> Via: SIP/2.0/WS an6ikqlgivd7.invalid;branch=z9hG4bK5845620
> Max-Forwards: 69
> To: <sip:bob at 93.187.138.214 <mailto:sip%3Abob at 93.187.138.214>>
> From: "Alice Test" <sip:alice at 93.187.138.214
> <mailto:sip%3Aalice at 93.187.138.214>>;tag=dt8iuu64l9
> Call-ID: bmaapitncfv1jnijrbcf
> CSeq: 7318 INVITE
> Contact: <sip:sbt6u2o1 at an6ikqlgivd7.invalid;transport=ws;ob>
> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE
> Content-Type: application/sdp
> Supported: path, outbound, gruu
> User-Agent: JsSIP 0.3.0
> Content-Length: 2967
>
> v=0
> o=- 1167703101330838157 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS 3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> m=audio 9496 RTP/SAVPF 111 103 0 8 106 105 13 126
> c=IN IP4 213.233.85.51
> a=rtcp:9496 IN IP4 213.233.85.51
> a=candidate:3511930567 1 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:3511930567 2 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:2681221687 1 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:2681221687 2 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:1343998067 1 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=candidate:1343998067 2 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=sendrecv
> a=rtcp-mux
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:uWFSQ5+41i2e12WFnGJKCfe+kuudHy0NurAaT8or
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:103 ISAC/16000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:231261060 cname:aZBL5jB9VQtchKUh
> a=ssrc:231261060 msid:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> e8c9eb9c-916e-4c30-884f-fd602b2d8c90
> a=ssrc:231261060 mslabel:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> a=ssrc:231261060 label:e8c9eb9c-916e-4c30-884f-fd602b2d8c90
> m=video 9496 RTP/SAVPF 100 116 117
> c=IN IP4 213.233.85.51
> a=rtcp:9496 IN IP4 213.233.85.51
> a=candidate:3511930567 1 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:3511930567 2 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:2681221687 1 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:2681221687 2 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:1343998067 1 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=candidate:1343998067 2 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:uWFSQ5+41i2e12WFnGJKCfe+kuudHy0NurAaT8or
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=ssrc:3207772497 cname:aZBL5jB9VQtchKUh
> a=ssrc:3207772497 msid:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> ec757148-040c-479f-adbe-f6bac271fbd6
> a=ssrc:3207772497 mslabel:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> a=ssrc:3207772497 label:ec757148-040c-479f-adbe-f6bac271fbd6
>
>
> AFTER mediaproxy-ng magic:
> 14(21473) DEBUG: rtpproxy-ng [rtpproxy.c:1333]: rtpp_function_call():
> proxy reply: d3:sdp3046:
> v=0
> o=- 1167703101330838157 2 IN IP4 93.187.138.214
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS 3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> m=audio 30408 RTP/SAVPF 111 103 0 8 106 105 13 126
> c=IN IP4 93.187.138.214
> a=candidate:3511930567 1 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:3511930567 2 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:2681221687 1 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:2681221687 2 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:1343998067 1 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=candidate:1343998067 2 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:audio
> a=sendrecv
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:uWFSQ5+41i2e12WFnGJKCfe+kuudHy0NurAaT8or
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:103 ISAC/16000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:231261060 cname:aZBL5jB9VQtchKUh
> a=ssrc:231261060 msid:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> e8c9eb9c-916e-4c30-884f-fd602b2d8c90
> a=ssrc:231261060 mslabel:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> a=ssrc:231261060 label:e8c9eb9c-916e-4c30-884f-fd602b2d8c90
> a=rtcp:30409
> a=candidate:8JGnwCAu0icAoJnr 1 UDP 2130706432 93.187.138.214 30408 typ host
> a=candidate:8JGnwCAu0icAoJnr 2 UDP 2130706431 93.187.138.214 30409 typ host
> m=video 30408 RTP/SAVPF 100 116 117
> c=IN IP4 93.187.138.214
> a=candidate:3511930567 1 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:3511930567 2 udp 2113937151 10.93.108.223 53310 typ host
> generation 0
> a=candidate:2681221687 1 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:2681221687 2 tcp 1509957375 10.93.108.223 0 typ host
> generation 0
> a=candidate:1343998067 1 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=candidate:1343998067 2 udp 1845501695 213.233.85.51 9496 typ srflx
> raddr 10.93.108.223 rport 53310 generation 0
> a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:video
> a=sendrecv
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:uWFSQ5+41i2e12WFnGJKCfe+kuudHy0NurAaT8or
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=ssrc:3207772497 cname:aZBL5jB9VQtchKUh
> a=ssrc:3207772497 msid:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> ec757148-040c-479f-adbe-f6bac271fbd6
> a=ssrc:3207772497 mslabel:3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
> a=ssrc:3207772497 label:ec757148-040c-479f-adbe-f6bac271fbd6
> a=rtcp:30409
> a=candidate:8JGnwCAu0icAoJnr 1 UDP 2130706432 93.187.138.214 30408 typ host
> a=candidate:8JGnwCAu0icAoJnr 2 UDP 2130706431 93.187.138.214 30409 typ host
>
>
>
> Between them, I have some strange logs in kamailio:
> 14(21473) ERROR: *** cfgtrace:
> c=[/usr/local/etc/kamailio/kamailio-test-media.cfg] l=878 a=25
> n=rtpproxy_manage
> 14(21473) DEBUG: <core> [parser/sdp/sdp_helpr_funcs.c:565]:
> extract_mediaip(): located IP address [127.0.0.1] in `o=' field
> 14(21473) DEBUG: <core> [parser/sdp/sdp_helpr_funcs.c:565]:
> extract_mediaip(): located IP address [213.233.85.51] in `c=' field
> 14(21473) DEBUG: <core> [parser/sdp/sdp.c:574]: parse_sdp_session():
> ignoring unknown type in a= line: `a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:audio
> ...............................................................
> 14(21473) DEBUG: <core> [parser/sdp/sdp.c:574]: parse_sdp_session():
> ignoring unknown type in a= line: `a=ssrc:3207772497
> label:ec757148-040c-479f-adbe-f6bac271fbd6
> '
> 14(21473) DEBUG: rtpproxy-ng [rtpproxy_funcs.c:148]:
> check_content_type(): type <application/sdp> found valid
> 14(21473) DEBUG: rtpproxy-ng [rtpproxy.c:1333]: rtpp_function_call():
> proxy reply: d3:sdp3046:v=0
> o=- 1167703101330838157 2 IN IP4 93.187.138.214
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS 3G8DreUBMks5DpcqNiiE5jXIC4GzIwfd7CUv
>
>
> Thank you very much for your help and for spending time debugging this
> error.
>
>
> Best regards,
> Mihai M
>
>
> On Wed, Feb 5, 2014 at 5:41 PM, Richard Fuchs <rfuchs at sipwise.com
> <mailto:rfuchs at sipwise.com>> wrote:
>
> Hey,
>
> If you're trying to connect two WebRTC endpoints with each, you don't
> need any of mediaproxy-ng's magic to get it working. All the previous
> replies were assuming that you were trying to connect a WebRTC endpoint
> with a non-WebRTC one, which is usually what people are trying to do.
>
> In your case, the flags "froc" in either direction should be sufficient
> to get the job done. If it still doesn't work, can you please post the
> rejected SDP body as it appears both on the sender's side and on the
> receiver's side (i.e. both before and after it went through MP-NG).
>
> cheers
>
>
> On 02/05/14 05:17, Mihai Marin wrote:
> > Hello,
> > Thank you for your detailed explication but I miss some information or
> > I'm unable to understand it properly. What I'm trying to do is to use
> > mediaproxy-ng as a turn server between 2 WebRTC endpoints (when at
> least
> > one is behind restrictive firewall). Trying to replicate what you
> > explained on my needs I tried:
> > $avp(rtpproxy_offer_flags) = "froc+SP";
> > $avp(rtpproxy_answer_flags) = "froc-SP";
> >
> > But, unfortunately, I have the same error. Sorry if the solution is
> > obvious but I can't find it.
> >
> > Thank you.
> >
> > Best regards,
> > Mihai M
> >
> >
> > On Tue, Feb 4, 2014 at 10:45 PM, Muhammad Shahzad
> <shaheryarkh at gmail.com <mailto:shaheryarkh at gmail.com>
> > <mailto:shaheryarkh at gmail.com <mailto:shaheryarkh at gmail.com>>> wrote:
> >
> > There are several problems that need to be addressed in your
> > kamailio.cfg but let me try to focus only on mediaprxoy-ng
> related ones.
> >
> > First instead of engaging mediaproxy in failure route, engage it
> > main route or branch route. Why wait for failure when we know call
> > will fail anyway if you try to call webrtc to sip or vice versa.
> >
> > Secondly you need to keep track of connection type of both caller
> > and callee and set appropriate mediaproxy-ng flags according
> to call
> > direction, e.g. call from webrtc to sip, or sip to webrtc or
> webrtc
> > to webrtc or sip to sip, each type of call needs different set of
> > flags for both rtpproxy_offer and rtpproxy_answer.
> >
> > How you do this, is pretty simple, to detect if caller is webrtc
> > endpoint you can use,
> >
> >
> > if ($avp(mline) =~ "SAVPF") {
> > # caller is a webrtc endpoint
> > };
> >
> > To check if callee is a webrtc endpoint, you can use,
> >
> > if ($(ru{uri.param,transport}) =~ "ws") {
> > # callee is a webrtc endpoint
> > };
> >
> > For testing purpose, i recommend you only use mediaproxy-ng for
> > bridging webrtc to sip or vice versa calls, i.e. if both endpoints
> > are using same transport (e.g. sip to sip or webrtc to webrtc
> calls)
> > then don't use mediaproxy-ng at all and allow endpoints to
> establish
> > media directly (that would work out the box at least for webrtc to
> > webrtc calls).
> >
> > Finally use correct flags for each type of call (i recommend doing
> > it in branch route), for example,
> >
> > For WebRTC to SIP call use flags (case-sensitive),
> >
> > $avp(rtpproxy_offer_flags) = "froc-sp";
> > $avp(rtpproxy_answer_flags) = "froc+SP";
> > rtpproxy_offer($avp(rtpproxy_offer_flags));
> >
> > For SIP to WebRTC call use flags (case-sensitive),
> >
> > $avp(rtpproxy_offer_flags) = "froc+SP";
> > $avp(rtpproxy_answer_flags) = "froc-sp";
> > rtpproxy_offer($avp(rtpproxy_offer_flags));
> >
> >
> > Then in reply route,
> >
> > rtpproxy_answer($avp(rtpproxy_answer_flags));
> >
> >
> > Remember, currently mediaproxy-ng does NOT support SRTP/DTLS,
> which
> > is required by firefox, so as result your webrtc endpoint MUST be
> > running on Chrome.
> >
> > Hope this helps.
> >
> > Thank you.
> >
> >
> >
> >
> > On Tue, Feb 4, 2014 at 3:28 PM, Mihai Marin
> <marinmihai at gmail.com <mailto:marinmihai at gmail.com>
> > <mailto:marinmihai at gmail.com <mailto:marinmihai at gmail.com>>>
> wrote:
> >
> > Hello,
> > Thank you for your support.
> >
> > Yes, I have the same error without video enabled. I have
> > attached the logs from jssip (with and without video support)
> > and logs from kamailio when trying a call with video support
> > enabled. The kamailio.cfg used is the same from my
> previous mail.
> >
> > I also tried with sipml5 and I have the same behavior.
> >
> > I'm stuck on this error and I think I'm looking in the wrong
> > direction.
> >
> > Thank you.
> >
> > Best regards,
> > Mihai M
> >
> >
> > On Tue, Feb 4, 2014 at 2:49 PM, Andrew Pogrebennyk
> > <apogrebennyk at sipwise.com
> <mailto:apogrebennyk at sipwise.com> <mailto:apogrebennyk at sipwise.com
> <mailto:apogrebennyk at sipwise.com>>> wrote:
> >
> > Hi,
> > could you please post also your Chrome js developer log?
> > Does the problem exist if you start the jssip clients
> > without video support?
> >
> > Andrew
> >
> > On 02/03/2014 12:00 PM, Mihai Marin wrote:
> > > Hello,
> > >
> > > Another weekend struggling to make a call from jssip to
> > another jssip
> > > behind firewall and I still receive 488 - Not Acceptable
> > Here. I tried
> > > all the ideas that I had/received without any success -
> > including catch
> > > 488 and re-invite.
> > > [...]
> > > What do I miss from my configuration?
> > >
> > > Thank you.
> > >
> > > Best regards,
> > > Mihai M
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> > mailing list
> > sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>
> > <mailto:sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>>
> >
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> > mailing list
> > sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>
> <mailto:sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>>
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> mailing list
> > sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>
> <mailto:sr-users at lists.sip-router.org
> <mailto:sr-users at lists.sip-router.org>>
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> list
> > sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 897 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140206/34b686ab/attachment.pgp>
More information about the sr-users
mailing list