Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind successfully but that's it.
Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio.
My config is located here: http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind successfully but that's it.
Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio.
My config is located here: http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
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
Hi Fred,
The forward path is like this: SipML5 -> [WSS (UDP/TLS/RTP/SAVPF)] -> Kamailio -> RTPEngine -> [SIP (RTP/AVP)] -> OpenMCU-ru
The return is opposite of this.
Call setup over SIP works fine from one end to the other and back.
Media i know is fine from SipML5 to Kamailio on the forward path over WSS. I do not see forward media going out of the machine. So maybe it's better termed that SIP is fine but RTP is broken in both directions. I see STUN negotiate successfully on this portion as well between SipML5 and Kamailio/RTPEngine.
The return path, media generated by OpenMCU-ru is sent back over via RTP/AVP to Kamailio/RTPEngine and that is where it gets lost.
Thanks, -Don
On Fri, Feb 6, 2015 at 5:47 AM, Fred Posner fred@palner.com wrote:
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind successfully but that's it.
Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio.
My config is located here: http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
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
I guess I should also note that I do have freeswitch on the machine and the contexts are there. It seems to work fine in echotest and conference modes. But I do have to caveat that I have not tried it recently with some recent config changes to handle the RTP/SAVPF to RTP/AVP rewriting - so that I cannot be sure of.
On Fri, Feb 6, 2015 at 5:47 AM, Fred Posner fred@palner.com wrote:
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind successfully but that's it.
Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio.
My config is located here: http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
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
I think i have similar problem last week with rtpengine deployment which was about 1-2 weeks old. There was no audio although the logs say that STUN bindings are successful from both side (SAVPF <-> AVP). One symptom of the problem is this log message,
-- rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 port 41198] SRTP output wanted, but no crypto suite was negotiated --
As a last resort before filing an official bug, i decided to upgrade RTPEngine which seems to have solved the problem. Interestingly both rtpengine deployments (the one causing no audio and the upgraded one) have the same version numbers (i.e. the output of "rtpengine -v"), so i can't actually pinpoint which revision has this problem and which one has solved this problem. Anyhow, can you also try with latest RTPEngine from official git repo and see if that solves the problem. The git commit number of working RTPEngine is 9a2da87f130ab3c1e21d9b593efec78a8eb7b3f3 (ah, i miss subversion which has more meaningful linear revision numbers ...).
For RTPEngine developers, can you guys add git revision string as extended version for "rtpengine -v" output? It may be the same way as kamailio does, e.g.
kamailio 4.2.2 (i386/linux) *6f7306*
This will tremendously help in tracking bugs and their fixes.
Thank you.
On Fri, Feb 6, 2015 at 5:07 PM, Don Fanning don@00100100.net wrote:
I guess I should also note that I do have freeswitch on the machine and the contexts are there. It seems to work fine in echotest and conference modes. But I do have to caveat that I have not tried it recently with some recent config changes to handle the RTP/SAVPF to RTP/AVP rewriting - so that I cannot be sure of.
On Fri, Feb 6, 2015 at 5:47 AM, Fred Posner fred@palner.com wrote:
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind successfully but that's it.
Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio.
My config is located here: http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
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
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
On 07/02/15 07:24 AM, Muhammad Shahzad wrote:
I think i have similar problem last week with rtpengine deployment which was about 1-2 weeks old. There was no audio although the logs say that STUN bindings are successful from both side (SAVPF <-> AVP). One symptom of the problem is this log message,
-- rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 port 41198] SRTP output wanted, but no crypto suite was negotiated --
As a last resort before filing an official bug, i decided to upgrade RTPEngine which seems to have solved the problem. Interestingly both rtpengine deployments (the one causing no audio and the upgraded one) have the same version numbers (i.e. the output of "rtpengine -v"), so i can't actually pinpoint which revision has this problem and which one has solved this problem. Anyhow, can you also try with latest RTPEngine from official git repo and see if that solves the problem. The git commit number of working RTPEngine is 9a2da87f130ab3c1e21d9b593efec78a8eb7b3f3 (ah, i miss subversion which has more meaningful linear revision numbers ...).
For RTPEngine developers, can you guys add git revision string as extended version for "rtpengine -v" output? It may be the same way as kamailio does, e.g.
kamailio 4.2.2 (i386/linux) *6f7306*
This will tremendously help in tracking bugs and their fixes.
It does, but it only works if the build environment has git available. Otherwise it has to be left out.
tabasco:~/src/ngcp-git/rtpengine/daemon(master)# ./rtpengine -v 3.3.0.0+0~mr3.8.0.0 git-master-9a2da87
cheers
I myself am using a git pull from master with this being the last (first) entry.
commit a0068f4f02c47af80685e9c0687c85e74b4e29af Author: Richard Fuchs rfuchs@sipwise.com Date: Thu Jan 8 10:47:52 2015 -0500
relax sdes key lifetime validation check
fixes #57
On Sat, Feb 7, 2015 at 5:53 AM, Richard Fuchs rfuchs@sipwise.com wrote:
On 07/02/15 07:24 AM, Muhammad Shahzad wrote:
I think i have similar problem last week with rtpengine deployment which was about 1-2 weeks old. There was no audio although the logs say that STUN bindings are successful from both side (SAVPF <-> AVP). One symptom of the problem is this log message,
-- rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 port 41198] SRTP output wanted, but no crypto suite was negotiated --
As a last resort before filing an official bug, i decided to upgrade RTPEngine which seems to have solved the problem. Interestingly both rtpengine deployments (the one causing no audio and the upgraded one) have the same version numbers (i.e. the output of "rtpengine -v"), so i can't actually pinpoint which revision has this problem and which one has solved this problem. Anyhow, can you also try with latest RTPEngine from official git repo and see if that solves the problem. The git commit number of working RTPEngine is 9a2da87f130ab3c1e21d9b593efec78a8eb7b3f3 (ah, i miss subversion which has more meaningful linear revision numbers ...).
For RTPEngine developers, can you guys add git revision string as extended version for "rtpengine -v" output? It may be the same way as kamailio does, e.g.
kamailio 4.2.2 (i386/linux) *6f7306*
This will tremendously help in tracking bugs and their fixes.
It does, but it only works if the build environment has git available. Otherwise it has to be left out.
tabasco:~/src/ngcp-git/rtpengine/daemon(master)# ./rtpengine -v 3.3.0.0+0~mr3.8.0.0 git-master-9a2da87
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
I just took Muhammad's advice and recompiled with a current git pull from master (37d98ad3ed3c29acfaa1f9a7d8a9404397dc442b) and can report success! I do recall seeing the exact same dtls log messages but did not focus on it.
On Sat, Feb 7, 2015 at 1:22 PM, Don Fanning don@00100100.net wrote:
I myself am using a git pull from master with this being the last (first) entry.
commit a0068f4f02c47af80685e9c0687c85e74b4e29af Author: Richard Fuchs rfuchs@sipwise.com Date: Thu Jan 8 10:47:52 2015 -0500
relax sdes key lifetime validation check fixes #57On Sat, Feb 7, 2015 at 5:53 AM, Richard Fuchs rfuchs@sipwise.com wrote:
On 07/02/15 07:24 AM, Muhammad Shahzad wrote:
I think i have similar problem last week with rtpengine deployment which was about 1-2 weeks old. There was no audio although the logs say that STUN bindings are successful from both side (SAVPF <-> AVP). One symptom of the problem is this log message,
-- rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 port 41198] SRTP output wanted, but no crypto suite was negotiated --
As a last resort before filing an official bug, i decided to upgrade RTPEngine which seems to have solved the problem. Interestingly both rtpengine deployments (the one causing no audio and the upgraded one) have the same version numbers (i.e. the output of "rtpengine -v"), so i can't actually pinpoint which revision has this problem and which one has solved this problem. Anyhow, can you also try with latest RTPEngine from official git repo and see if that solves the problem. The git commit number of working RTPEngine is 9a2da87f130ab3c1e21d9b593efec78a8eb7b3f3 (ah, i miss subversion which has more meaningful linear revision numbers ...).
For RTPEngine developers, can you guys add git revision string as extended version for "rtpengine -v" output? It may be the same way as kamailio does, e.g.
kamailio 4.2.2 (i386/linux) *6f7306*
This will tremendously help in tracking bugs and their fixes.
It does, but it only works if the build environment has git available. Otherwise it has to be left out.
tabasco:~/src/ngcp-git/rtpengine/daemon(master)# ./rtpengine -v 3.3.0.0+0~mr3.8.0.0 git-master-9a2da87
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
-
Don Fanning -
Fred Posner -
Muhammad Shahzad -
Richard Fuchs