Re: [SR-Users] kamailio + rtpengine: damaged media (Red Hat Linux)
2014-06-10 20:09 GMT+04:00 Richard Fuchs rfu...@sipwise.com: On 06/10/14 05:41, Alexey Rybalko wrote: As the problem occurs on encrypted RTP (DTLS-SRTP) only, I suppose that 'glitched' media is wrong encrypted or unencrypted payload for some unknown reason. Turns out it was a silly typo that sneaked in recently. Please update from rtpengine master and try with that. I confirm it works now through the kernel module. Many thanks for fixing it quickly! regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio + rtpengine: damaged media (Red Hat Linux)
Hello! 2014-06-09 19:06 GMT+04:00 Richard Fuchs rfu...@sipwise.com: Hard to tell what the problem is without looking at the RTP traffic. The log looks fine. The delay you mentioned could indicate that it might be the kernel module that's misbehaving. Perhaps you can try the same thing again, but without the iptables rules installed (and/or without the kernel module loaded), and see if that makes a difference. Richard, thank you. I believe the problem is related to the kernel module. Media flows smoothly without it. (Just have tried with no module loaded and no iptables' rules). I have sent rtp dump to your email. May be you would have the time to check it. As the problem occurs on encrypted RTP (DTLS-SRTP) only, I suppose that 'glitched' media is wrong encrypted or unencrypted payload for some unknown reason. That's curious, perhaps something went wrong when compiling the kernel module then? I have to admit that I have zero experience with rtpengine on Red Hat myself. However there were no errors during the module compilation: *$ MEDIAPROXY_VERSION=\3.3.0.0\ makemake -C /lib/modules/2.6.32-358.el6.x86_64/build M=/usr/src/rtpengine/kernel-module O=/lib/modules/2.6.32-358.el6.x86_64/build modulesmake[1]: Entering directory `/usr/src/kernels/2.6.32-358.el6.x86_64' CC [M] /usr/src/rtpengine/kernel-module/xt_MEDIAPROXY.o Building modules, stage 2. MODPOST 1 modules CC /usr/src/rtpengine/kernel-module/xt_MEDIAPROXY.mod.o LD [M] /usr/src/rtpengine/kernel-module/xt_MEDIAPROXY.ko.unsigned NO SIGN [M] /usr/src/rtpengine/kernel-module/xt_MEDIAPROXY.ko make[1]: Leaving directory `/usr/src/kernels/2.6.32-358.el6.x86_64'* regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] [rtpengine] No media from WebRTC UA
Hello! During a call from classical SIP softphone to WebRTC there's no media from the browser (Mozilla, the same result is for Chrome). In case of a call from the browser to the softphone there's media flow from both sides. The snippets from kamailio.cfg related to the problem case (SIP--WebRTC) are below. OFFER: $var(rtpp_flags) = trust-address symmetric replace-origin replace-session-connection; $var(rtpp_flags) = $var(rtpp_flags) + ICE=force; $var(rtpp_flags) = $var(rtpp_flags) + RTP/SAVPF; rtpengine_offer($var(rtpp_flags)); ANSWER: $var(rtpp_flags) = trust-address symmetric replace-origin replace-session-connection; $var(rtpp_flags) = $var(rtpp_flags) + ICE=remove; $var(rtpp_flags) = $var(rtpp_flags) + RTP/AVP; rtp.log is attached. Any help on this issue would be very appreciated. with best regards, Alexey rtp.log Description: Binary data ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [rtpengine] No media from WebRTC UA
Hi Richard! Just have tried an outgoing call to Chrome and Opera and it works fine. Thank you for the clarification regarding ICE! Dump files and rtp logs (Firefox, Chrome) I've sent to your email. My little investigation brings the following : - browsers (Firefox, Chrome and Opera) don't use a=ice-lite in their SDP; - Chrome and Opera take the role of ICE-CONTROLLING and provide USE-CANDIDATE for the ICE-Lite peer (mediaproxy) for both offer and answer cases; - Firefox takes the role of ICE-CONTROLLING and provide USE-CANDIDATE in case of the offer while it takes the role of ICE-CONTROLLED during the answer; /means no media from WebRTC UA in the latter case/ Some other remarks. During the call from Fire I saw a lot of SRTP output wanted, but no crypto suite was negotiated messages from rtpengine. However DTLS is finally was established. Is that one more issue of Firefox? Looking in STUN section of the dump files I wonder why Chrome use more than 10 binding request (USE-CANDIDATE) for each candidate while Mozilla does it just once. regards, Alexey 2014-05-16 14:53 GMT+04:00 Richard Fuchs rfu...@sipwise.com: There's nothing wrong with the SDP bodies that I can see. I recall that Firefox had or still has a problem with ICE role switching when ice-lite is offered. It never completes ICE negotiation (never sends an STUN packet with use candidate) and so never starts DTLS handshake. You can confirm that by doing a packet capture including the RTP ports and inspecting the STUN packets. Chrome shouldn't have that problem though, perhaps do another test run with it? You can send those capture files to me if you'd like me to have a look. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [rtpengine] No media from WebRTC UA
Hi, Alex! I'm not experienced in Kamailio, but that code works well :) Is there any difference between feeding the function with variable (AVP?) or string literal? best regards, Alexey 2014-05-16 16:09 GMT+04:00 Alex Balashov abalas...@evaristesys.com: Hello Alexey, I am uncertain as to whether rtpengine_offer()/answer() support pseudovariable arguments. But if they do, you'll need to wrap them in a string literal: rtpengine_offer($var(rtpp_flags)); If they don't support PV arguments at all, you may be stuck with having to provide a static argument: rtpengine_offer(trust-address symmetric replace-origin replace-session-connection ICE=force RTP/SAVPF); -- Alex ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No secure attributes from rtpengine in SRTP/RTP bridge mode
Hello! 2014-04-27 2:42 GMT+04:00 Richard Fuchs rfu...@sipwise.com: Please try again with ICE=force instead of force_relay, or (more conservatively) ICE=remove in the offer and ICE=force in the answer. You will probably also need the option rtcp-mux-demux in the offer, as your non-RTC endpoint doesn't support rtcp-mux and Chrome isn't likely to be happy with its rtcp-mux offer being declined. Richard, I've tried the options you had recommended: ICE=force in offer and answer (ICE=remove in the offer works well too). The audio call was negotiated and traversed. Thank you very much! 2014-04-27 9:27 GMT+04:00 Juha Heinanen j...@tutpro.com: Alexey Rybalko writes: There is no such attribute in SDP payload from the latest Mozilla (v.29). there *is* a=setup:actpass in above. Juha, thanks for the hint! Yesterday it was too late at night, I had missed that string J Today made a call from Mozilla to softphone. regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No secure attributes from rtpengine in SRTP/RTP bridge mode
Hi, Richard! Thank you for quick feedback! There is no such attribute in SDP payload from the latest Mozilla (v.29). v=0 o=Mozilla-SIPUA-29.0 371 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:083b4837 a=ice-pwd:dac461d48770be5e1dae6c450e144bf3 a=fingerprint:sha-256 C3:AA:DB:75:D7:60:FC:B6:94:A7:81:4F:74:A2:FF:44:4B:17:AE:D3:64:37:37:D1:AC:1A:F5:D4:86:1E:4F:7A m=audio 52775 RTP/SAVPF 109 0 8 101 c=IN IP4 192.168.0.101 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=setup:actpass a=candidate:0 1 UDP 2130444543 192.168.0.101 52775 typ host a=candidate:0 2 UDP 2130444542 192.168.0.101 63139 typ host a=rtcp-mux But it presents in SDP from Chrome (v.34) v=0 o=- 1634904605592690072 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS C8ATLgPd2jIcc5q799L9XU3rTROMajedYbdI m=audio 51817 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.0.101 a=rtcp:51817 IN IP4 192.168.0.101 a=candidate:3350409123 1 udp 2122260223 192.168.0.101 51817 typ host generation 0 a=candidate:3350409123 2 udp 2122260223 192.168.0.101 51817 typ host generation 0 a=candidate:2301678419 1 tcp 1518280447 192.168.0.101 0 typ host generation 0 a=candidate:2301678419 2 tcp 1518280447 192.168.0.101 0 typ host generation 0 a=ice-ufrag:WyHALLFH6CaQmCIA a=ice-pwd:9BMkH9d7D9pfSjZmLSkunxrW a=ice-options:google-ice a=fingerprint:sha-256 46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22 *a=setup:actpass* a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:cDj0wVDUUZ/1etNd9MFQjeqwn/ii3RsxQLraXUln a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:KKzZx0iwM2udfGNv+pBoB/BDVBvsFsMcQczVZDOQ No success for both browsers. It's should be noticed that Chrome provides both SDES (crypto) and DTLS (fingerprint+setup:actpass) attibutes (does DTLS have priority in a such case?). However rtpengine doesn't provide such SRTP data. May be any suggestions? kind regards, Alexey 2014-04-25 18:02 GMT+04:00 Richard Fuchs rfu...@sipwise.com: Hi, Can you check if the original offer contains an a=setup:actpass attribute? I remember Firefox having a problem with this in some version. This attribute is required for DTLS-SRTP and Firefox was not sending it. It's fixed in later versions. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No secure attributes from rtpengine in SRTP/RTP bridge mode
Failed to set remote answer sdp: Called with a SDP without crypto enabled (Chrome) RTPEngine log is attached. regards, /A 2014-04-27 2:09 GMT+04:00 Richard Fuchs rfu...@sipwise.com: On 04/26/14 17:32, Alexey Rybalko wrote: No success for both browsers. It's should be noticed that Chrome provides both SDES (crypto) and DTLS (fingerprint+setup:actpass) attibutes (does DTLS have priority in a such case?). However rtpengine doesn't provide such SRTP data. May be any suggestions? It can't work with Firefox if it doesn't use the a=setup attribute, but it should work with Chrome. If it doesn't, please provide the complete syslog output from rtpengine for such a call. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users rtp.log Description: Binary data ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] No secure attributes from rtpengine in SRTP/RTP bridge mode
Hello!
I have been experimenting with drop-in replacement of old rtpproxy-ng
module with new rtpengine. Wondering what is wrong in my configuration:
there are no security attributes in rtpengine answer on RTP/SAVPF offer.
Neither fingerprint (DTLS) nor crypto (SDES).
I used Firefox 29 during this test.
1. Here's original offer :
INVITE sip:user5@. SIP/2.0
Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444
Max-Forwards: 69
To: sip:user5@..
From: user4 sip:user4@..;tag=dvf1co8urv
Call-ID: phmq9o62cv21timhfnpf
CSeq: 4233 INVITE
Proxy-Authorization: Digest algorithm=MD5, username=user4,
realm=.., nonce=U1o8H1NaOvP3wxegsYCKOJX7S7DV/r1N,
uri=sip:user5@..,
response=44e7b16c55d3237f63e04b3c0b194f45
Contact: sip:user4@
..;gr=urn:uuid:c193bcd4-aa2e-47ef-a106-22e60f5fde9e;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 607
v=0
o=Mozilla-SIPUA-29.0 15825 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:2102f082
a=ice-pwd:8733e5248a7fb087b40ea45b3ca6f634
*a=fingerprint:sha-256
32:AA:85:DB:D8:3C:E6:C3:46:07:11:9E:9F:54:B9:42:7F:5C:37:5F:9D:D1:AD:19:22:A3:AC:9C:6C:A5:A6:CD*
m=audio 62290 *RTP/SAVPF* 109 0 8 101
c=IN IP4 .
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
.
Here's the snippet for translation SRTP-RTP:
rtpengine_offer(force trust-address symmetric replace-origin
replace-session-connection ICE=force_relay *RTP/AVP*);
2. Here's final answer (from rtpengine):
SIP/2.0 200 OK
Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444
Record-Route: sip:..;lr;nat=yes
Record-Route: sip:.:15060;transport=udp;lr;ovid=3207d8cd
Record-Route: sip:95f6551e81@...:10080;transport=ws;lr;ovid=3207d8cd
Contact: sip:user5@..1:49362
To: sip:user5@.;tag=4d436110
From: user4sip:user4@;tag=dvf1co8urv
Call-ID: phmq9o62cv21timhfnpf
CSeq: 4233 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.4 stamp 70864
Content-Length: 432
v=0
o=- 1398422264455879 3 IN IP4 .
s=X-Lite 4 release 4.5.4 stamp 70864
c=IN IP4 ..
t=0 0
m=audio 30002 *RTP/SAVPF* 109 0 8 101
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:30003
Error from JsSIP:
{name: INTERNAL_ERROR, message: Could not negotiate answer SDP; cause =
*NO_DTLS_FINGERPRINT*, __exposedProps__: Object}
Here's the snippet for translation RTP-SRTP:
rtpengine_answer(force trust-address symmetric replace-origin
replace-session-connection rtcp-mux-demux ICE=force *RTP/SAVPF* );
There was another test with Chrome 34 with the same result.
Offer:
INVITE sip:user5@ SIP/2.0 Via: SIP/2.0/WS
ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 Max-Forwards: 69 To:
sip:user5@
From: user4 sip:user4@;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb
CSeq: 957 INVITE Proxy-Authorization: Digest algorithm=MD5,
username=user4, realm=., nonce=U1pKZ1NaSTtLo2vjK9TGyBu6Axb+EtyN,
uri=sip:user5@..., response=98f6275d3636664b611ff1411af982af Contact:
sip:user4@;gr=urn:uuid:8cd4e797-7314-485f-b191-4a15a6581c42;ob Allow:
ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp
Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length:
1586 v=0 o=- 7510391807340598328 2 IN IP4 127.0.0.1 s=- t=0 0
a=group:BUNDLE audio a=msid-semantic: WMS
ErnvtgCDe9aR6LWxNRgT83r0mxtyAV87LUxT m=audio 51672 *RTP/SAVPF *111 103 104
0 8 106 105 13 126 c=IN IP4 10.61.2.151 a=rtcp:51672 IN IP4 ..
a=ice-ufrag:EMn7uHfSS7ulRGU2 a=ice-pwd:FskTdhj7qT6ELP7uTIb+gquQ
a=ice-options:google-ice *a=fingerprint:sha-256
46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22*a=setup:actpass
a=mid:audio a=extmap:1
urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux *a=crypto:0
AES_CM_128_HMAC_SHA1_32 inline:R6qaiU7Cm471zNF6f3Q87TyXbHjEt/VhLgUgY2ZZ
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:lch+mfN/hi9QmseWu+ss1M2vA8mwRh8GQYChaJvc* a=rtpmap:111 opus/48000/2
Answer:
SIP/2.0 200 OK Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236
Record-Route: sip:...;lr;nat=yes Record-Route:
sip::15060;transport=udp;lr;ovid=3207d8cd Record-Route:
sip:712a450958@...:10080;transport=ws;lr;ovid=3207d8cd Contact:
sip:user5@10.61.2.151:49362 To: sip:user5@;tag=b8c8ee57 From:
user4sip:user4@...;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq:
957 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces,
eventlist User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432
v=0 o=- 1398425917116411 3 IN IP4 ... s=X-Lite 4
[SR-Users] Kamailio as a credit control application (RFC 4006)
Hi! Does Kamailio has features of CCA for usage outside a compiled code (cdp_avp)? There's a module *ims_ro_interface *listed at http://www.kamailio.org/w/2013/05/ims-kamailio/#more-1664 What status does it have? regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] mediaproxy-ng: error rewriting SDP during a video call
Hi! Few days ago I was lucky to establish calls between Chrome and SIP UA. Thanks to new rtpproxy developers! That was for audio only because many UAs lack for VP8 support. To verify a video I tried to involve Jitsi into the tests. Mediaproxy can't rewrite answer SDP from Jitsi. Probably the issue is related to SDP but I'm not sure to which part in particularly. *Original offer SDP (ICE attributes are skipped) * v=0 o=- 6784329718950523193 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP m=audio 59996 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 c=IN IP4 10.xx.xx.xx a=rtcp:59996 IN IP4 10.xx.xx.xx a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VZhAArr96fxUtzvuNPpE+OSa09wchsmTD9pdrdBv a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 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:579751405 cname:n2gMzhLDYdqpWwP7 a=ssrc:579751405 msid:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPa0 a=ssrc:579751405 mslabel:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP a=ssrc:579751405 label:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPa0 m=video 59996 RTP/SAVPF 100 116 117 c=IN IP4 10.xx.xx.xx a=rtcp:59996 IN IP4 10.xx.xx.xx ... a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=sendrecv a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VZhAArr96fxUtzvuNPpE+OSa09wchsmTD9pdrdBv a=rtpmap:100 VP8/9 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 goog-remb a=rtpmap:116 red/9 a=rtpmap:117 ulpfec/9 a=ssrc:1190198390 cname:n2gMzhLDYdqpWwP7 a=ssrc:1190198390 msid:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPv0 a=ssrc:1190198390 mslabel:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP a=ssrc:1190198390 label:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPv0 *Original answer SDP* v=0 o=user2 0 0 IN IP4 10.xx.xx.xx s=- c=IN IP4 10.xx.xx.xx t=0 0 m=audio 5052 RTP/AVP 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 m=video 5054 RTP/AVP 100 a=rtpmap:100 VP8/9 *Mediaproxy logs* (for answer SDP) mediaproxy-ng[14907]: Got valid command from 127.0.0.1:38310: answer mediaproxy-ng[14907]: [7dmpb0qdja07p87pa1ji - ] Got LOOKUP, but no usable callstreams found mediaproxy-ng[14907]: Error rewriting SDP mediaproxy-ng[14907]: Protocol error in packet from 127.0.0.1:38310: Error rewriting SDP:... However the *audio calls work well*. Here's the answer SDP from Jitsi: v=0 o=user2 0 0 IN IP4 10.xx.xx.xx s=- c=IN IP4 10.xx.xx.xx t=0 0 m=audio 5048 RTP/AVP 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 Comments and suggestions would be greatly appreciated. regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] mediaproxy-ng: error rewriting SDP during a video call
Richard, thank you for the quick response! Waiting for the fix. best regards, Alexey 01.08.2013 17:45 пользователь Richard Fuchs rfu...@sipwise.com написал: On 08/01/13 09:10, Alexey Rybalko wrote: Hi! Few days ago I was lucky to establish calls between Chrome and SIP UA. Thanks to new rtpproxy developers! That was for audio only because many UAs lack for VP8 support. To verify a video I tried to involve Jitsi into the tests. Mediaproxy can't rewrite answer SDP from Jitsi. Probably the issue is related to SDP but I'm not sure to which part in particularly. Hi, This is a known issue with mediaproxy-ng. The problem is that Chrome multiplexes both media streams on the same port and mediaproxy-ng currently doesn't handle this case correctly. I'm working on a fix, but it's a major change and will take a little while. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy-ng doesn't work with mediaproxy-ng
Hi!
Just my two cents. What parameters do you use for mediaproxy? Smth like
RUN_MEDIAPROXY=yes
LISTEN_UDP=127.0.0.1:
LISTEN_NG=127.0.0.1:11123
*ADDRESS=XXX.XXX.XXX.XXX*
?
regards,
Alexey
2013/7/26 Khue Nguyen Minh khu...@vega.com.vn
Thanks Richard. Now, rtpproxy-ng can work with mediaproxy-ng. But, I have
some error when run it. After rtpproxy-ng send SDP information to
mediaproxy-ng, it receive wrong IP. please see follow log:
Got valid command from 127.0.0.1:60340: offer - { sdp:
v=0#015#012o=doubango 1983 678901 IN IP4 10.0.0.19#015#012s=-#015#012c=IN
IP4 10.0.0.19#015#012t=0 0#015#012a=tcap:1 RTP/AVPF#015#012m=audio 18876
RTP/AVP 0 8 101#015#012a=ptime:20#015#012a=silenceSupp:off - - -
-#015#012a=rtpmap:0 PCMU/8000/1#015#012a=rtpmap:8
PCMA/8000/1#015#012a=rtpmap:101 telephone-event/8000/1#015#012a=fmtp:101
0-16#015#012a=pcfg:1
t=1#015#012a=sendrecv#015#012a=rtcp-mux#015#012a=ssrc:434299437
cname:ldjWoB60jbyQlR6e#015#012a=ssrc:434299437
mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2#015#012a=ssrc:434299437
label:Doubango#015#012m=text 2306 RTP/AVP 124 123#015#012a=rtpmap:124
t140/1000#015#012a=fmtp:124 cps=30#015#012a=rtpmap:123
red/1000#015#012a=fmtp:123 124/124/124/124#015#012a=pcfg:1
t=1#015#012a=sendrecv#015#012a=rtcp-mux#015#012, flags: [ force,
auto-bridge ], replace: [ origin ], call-id:
066cf8f8-fec9-7441-32f1-211298ff1715, received-from: [ IP4, x.x.x.x
], from-tag: 15056954, command: offer }
[066cf8f8-fec9-7441-32f1-211298ff1715] Creating new call
Returning to SIP proxy: d3:sdp1022:v=0#015#012o=doubango 1983 678901 IN
IP4 0.0.0.0#015#012s=-#015#012c=IN IP4 *0.0.0.0*#015#012t=0
0#015#012a=tcap:1 RTP/AVPF#015#012a=ice-lite#015#012m=audio 4 RTP/AVP 0
8 101#015#012a=ptime:20#015#012a=silenceSupp:off - - - -#015#012a=rtpmap:0
PCMU/8000/1#015#012a=rtpmap:8 PCMA/8000/1#015#012a=rtpmap:101
telephone-event/8000/1#015#012a=fmtp:101 0-16#015#012a=pcfg:1
t=1#015#012a=sendrecv#015#012a=ssrc:434299437
cname:ldjWoB60jbyQlR6e#015#012a=ssrc:434299437
mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2#015#012a=ssrc:434299437
label:Doubango#015#012a=rtcp:40001#015#012a=ice-ufrag:7YFi3AH6#015#012a=ice-pwd:oZwjQ1y9WJbLLI6mjWMSOOSVMvdQ#015#012a=candidate:rfWFb7Vp3QisWSvf
1 UDP 2130706432 0.0.0.0 4 typ host#015#012a=candidate:rfWFb7Vp3QisWSvf
2 UDP 2130706431 0.0.0.0 40001 typ host#015#012m=text 40004 RTP/AVP 124
123#015#012a=rtpmap:124 t140/1000#015#012a=fmtp:124
cps=30#015#012a=rtpmap:123 red/1000#015#012a=fmtp:123
124/124/124/124#015#012a=pcfg:1
t=1#015#012a=sendrecv#015#012a=rtcp:40005#015#012a=ice-ufrag:4SUsQrLE#015#012a=ice-pwd:qhLmZW7KFb6W7QOVFEOraZfcQaCG#015#012a=candidate:rfWFb7Vp3QisWSvf
1 UDP 2130706432 0.0.0.0 40004 typ host#015#012a=candidate:rfWFb7Vp3QisWSvf
2 UDP 2130706431* 0.0.0.0 *40005 typ host#015#0126:result2:oke
And in SIP message I capture from network, IP address in SDP body become
0.0.0.0. The call setup success but I don't hear anything. When I hangup
this call, module rtpproxy-ng segment fault, and this is call stack:
#0 bencode_string (bencbuf=0x7fffb6206760, msg=0x7fd617cb13a0,
op=OP_DELETE, flags_str=0x7fd617cd5f90 fox, body_out=0x0) at bencode.h:349
#1 bencode_list_add_string (bencbuf=0x7fffb6206760, msg=0x7fd617cb13a0,
op=OP_DELETE, flags_str=0x7fd617cd5f90 fox, body_out=0x0) at bencode.h:407
#2 rtpp_function_call (bencbuf=0x7fffb6206760, msg=0x7fd617cb13a0,
op=OP_DELETE, flags_str=0x7fd617cd5f90 fox, body_out=0x0) at
rtpproxy.c:1191
Please help me fix it.
Thanks
Khue.
2013/7/24 Richard Fuchs rfu...@sipwise.com
On 07/24/13 05:45, Khue Nguyen Minh wrote:
Hi all,
I am using rtpproxy-ng to control mediaproxy-ng. I was install and
config follow this guide:
https://github.com/sipwise/mediaproxy-ng
when I run kamailio with rtpproxy-ng module and mediaproxy-ng I got
error:
mediaproxy-ng[25216]: Failed to properly parse UDP command line '30514_2
d7:command4:pinge' from 127.0.0.1:54621 http://127.0.0.1:54621, using
fallback RE
ERROR: rtpproxy-ng [rtpproxy.c:1381]: rtpp_test(): proxy responded with
invalid response
As a quick guess, I'd say that you used the -u option instead of the -n
option (or --listen-udp instead of --listen-ng). Substitute one for the
other and it should work.
cheers
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[SR-Users] Empty location table
Hello! /kamilio 4.1.0/ Can anyone suggest why location table is empty despite that user registration works? save(location) return positive value. kamctl shows online users but if I look into kamailio.location it's empty anytime. # - registrar params - modparam(registrar, method_filtering, 0) modparam(registrar, append_branches, 1) modparam(registrar, max_contacts, 10) modparam(registrar, max_expires, 3600) ... Other strange the strange thing is that $rU is null. regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Empty location table
Sorry. My bad :/ There was a mistake in define WITH_US*E*RLOC. Just have found it hours later. cheers /A 2013/7/20 Alexey Rybalko alexey.ryba...@gmail.com Hello! /kamilio 4.1.0/ Can anyone suggest why location table is empty despite that user registration works? save(location) return positive value. kamctl shows online users but if I look into kamailio.location it's empty anytime. # - registrar params - modparam(registrar, method_filtering, 0) modparam(registrar, append_branches, 1) modparam(registrar, max_contacts, 10) modparam(registrar, max_expires, 3600) ... Other strange the strange thing is that $rU is null. regards, Alexey ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] mediaproxy-ng documentation available
Hi!
Just suggest someone already tried mediaproxy-ng with conversion RTP/SRTP.
Few examples of options' usage would be very appreciated! May authors bring
them into the tutorial?
E.g. caller invokes RTP/SAVPF profile (SIP over WS), but calle supports
RTP/AVP only. During the simple tests I've put *
rtpproxy_offer(spFRWOCII1-)* into the INVITE route. However it doesn't
work as was expected: mediaproxy offers the same SDP for callee. SIP 488
(Not Acceptable) as a result for caller.
Log events:
*[kamailio]*
...
ERROR: rtpproxy-ng [rtpproxy.c:1229]: unknown option `s'
...
ERROR: rtpproxy-ng [rtpproxy.c:1318]: proxy replied with error: Unknown
call-id
*[mediaproxy-ng]*
...
mediaproxy-ng[1507]: Got valid command from 127.0.0.1:57782: answer - {
sdp: v=0#015#012o=user2 0 0 IN IP4 10.61.24.86#015#012s=-#015#012c=IN
IP4 10.61.24.86#015#012t=0 0#015#012m=audio 5008 RTP/SAVPF 0 8
126#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:126
telephone-event/8000#015#012a=direction:active#015#012, ICE: remove,
flags: [ force, trust-address, symmetric ], replace: [ origin,
session-connection ], call-id: 5pl3eepicm9un6a3ir8e, via-branch:
z9hG4bK6568.ba6f970a58580cf0ae892a1a7b5aa09d.0, received-from: [ IP4,
127.0.0.1 ], from-tag: iqqqf0ucto, to-tag:
19CE5B50-51E7D8F60001641A-788B8700, command: answer }
mediaproxy-ng[1507]: Protocol error in packet from 127.0.0.1:57782: Unknown
call-id [d3:sdp202:v=0#015#012o=user2 0 0 IN IP4
10.61.24.86#015#012s=-#015#012c=IN IP4 10.61.24.86#015#012t=0
0#015#012m=audio 5008 RTP/SAVPF 0 8 126#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:126
telephone-event/8000#015#012a=direction:active#015#0123:ICE6:remove5:flagsl5:force13:trust-address9:symmetrice7:replacel6:origin18:session-connectione7:call-id20:5pl3eepicm9un6a3ir8e10:via-branch46:z9hG4bK6568.ba6f970a58580cf0ae892a1a7b5aa09d.013:received-froml3:IP49:127.0.0.1e8:from-tag10:iqqqf0ucto6:to-tag34:19CE5B50-51E7D8F60001641A-788B87007:command6:answere]
mediaproxy-ng[1507]: Returning to SIP proxy:
d6:result5:error12:error-reason15:Unknown call-ide
regards,
/A
2013/7/14 Richard Fuchs rfu...@sipwise.com
On 07/14/13 08:59, Alexey Rybalko wrote:
Regarding the mediaproxy-ng documentation special features can't be
invoked without usage of 'ng' protocol provided by rptmediaproxy-ng.
Unfortunately there is no info about rtpproxy-ng module itself. Haven't
found it at Sipwise GitHub site.
You can get it either as a patch file here:
https://github.com/sipwise/kamailio/blob/master/debian/patches/sipwise/rtproxy-ng.patch
Or by checking out the branch rfuchs/rtpproxy-ng in the regular Kamailio
repository.
cheers
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Re: [SR-Users] mediaproxy-ng documentation available
Richard, to be frank, I tried Sipwise's distribution of Kamailio (NGCP 2.8). Thanks for configured distro image as well :) Spending some time with tracing the config file brought the call stack: ROUTE_INVITE -...-ROUTE_BRANCH_ACC_RTP. However I've added sp flags into rtpproxy_offer function among other flags into ROUTE_BRANCH_ACC_RTP. That was the shortest way to figure out how mediaproxy works with media translation feature J Other reason for using the prepared distro is that I'm completely new to the Kamailio scripting and there are no other live examples for mediaproxy-ng usage. Regarding s flag..Hm...Just haven't thought that version of rtpproxy-ng might be out of date... Seems that I need to compile module from your Git branch. /Alexey 2013/7/18 Richard Fuchs rfu...@sipwise.com Hi, On 07/18/13 08:48, Alexey Rybalko wrote: Just suggest someone already tried mediaproxy-ng with conversion RTP/SRTP. Few examples of options' usage would be very appreciated! May authors bring them into the tutorial? E.g. caller invokes RTP/SAVPF profile (SIP over WS), but calle supports RTP/AVP only. During the simple tests I've put /rtpproxy_offer(spFRWOCII1-)**/ into the INVITE route. However it doesn't work as was expected: mediaproxy offers the same SDP for callee. SIP 488 (Not Acceptable) as a result for caller. Your usage is correct, but it doesn't seem to match the log lines you posted (those are for an answer, not an offer). Where exactly did you get the module from? Did you perhaps grab an older version, one that doesn't have the 's' flag implemented yet? cheers __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] mediaproxy-ng documentation available
Hi! Richard, Juha, thanks for the hint. Have fetched and rendered the module documentation. Now playing with translation between SRTP and RTP through mediaproxy: not trivial to focus on proper routes inside configuration file. I'm new to the Kamailio/OpenSER. Hope for the success J best regards, Alexey 2013/7/14 Richard Fuchs rfu...@sipwise.com On 07/14/13 08:59, Alexey Rybalko wrote: Regarding the mediaproxy-ng documentation special features can't be invoked without usage of 'ng' protocol provided by rptmediaproxy-ng. Unfortunately there is no info about rtpproxy-ng module itself. Haven't found it at Sipwise GitHub site. You can get it either as a patch file here: https://github.com/sipwise/kamailio/blob/master/debian/patches/sipwise/rtproxy-ng.patch Or by checking out the branch rfuchs/rtpproxy-ng in the regular Kamailio repository. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] mediaproxy-ng documentation available
Hi! Question regarding Kamalio rtpproxy-ng module for mediaproxy-ng daemon.* * * On Tue, Apr 2, 2013 at 7:05 PM, Richard Fuchs rfuchs at sipwise.com wrote: *The newest version supports a new control protocol, which is facilitated through a new Kamailio module rtpproxy-ng. For now, this module is a drop-in replacement for the old rtpproxy module and supports the same stuff, plus some ICE options. It's not included in the regular Kamailio git tree yet, but soon will be. In the meantime, the module is available from our github Kamailio tree. Regarding the mediaproxy-ng documentation special features can't be invoked without usage of 'ng' protocol provided by rptmediaproxy-ng. Unfortunately there is no info about rtpproxy-ng module itself. Haven't found it at Sipwise GitHub site. regards, Alexey 2013/7/11 Richard Fuchs rfu...@sipwise.com On 07/11/13 13:04, Juha Heinanen wrote: regarding r flag, if sip ua is behind nat, how can ip address in sdp be trusted, because source address of rtp packets does not match the one in sdp? Mediaproxy-ng pays attention to the source address of incoming packets and adjusts the forwarding address of the other stream direction accordingly (but only in the initial stage of a call to limit abuse). cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
