One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true)
We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. Call: Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. RTP streams, as negotiated (SDP): Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) RTP streams, as sent/receiced: FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] All this from Wireshark trace. I tried the following, with no success: 1. Added to MyFSApp.xml and dialplan\public.xml: <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Restarted, FS, same result. 2. Commented out <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="use-rtp-timer" value="true"/> Restarted, FS, same result. 3. Started FS with -nonat, then nonatmap Same result. Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old. What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? Thanks, A.C.
Can you supply a pastebin of adebug log of the call? Steve on iPhone On 12 Jul 2012, at 21:39, Adelia C. <hexade@hotmail.com> wrote:
We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time.
Call:
Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee.
RTP streams, as negotiated (SDP):
Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing)
Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP)
RTP streams, as sent/receiced:
FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.]
All this from Wireshark trace.
I tried the following, with no success:
1. Added to MyFSApp.xml and dialplan\public.xml: <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Restarted, FS, same result.
2. Commented out <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="use-rtp-timer" value="true"/> Restarted, FS, same result.
3. Started FS with -nonat, then nonatmap Same result.
Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old.
What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug?
Thanks, A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Thank you Steve - here it is - http://pastebin.freeswitch.org/19510. Fresh call, new ports. Notice this line, this is what I want to stop: 2012-07-13 10:18:42.935692 [INFO] switch_rtp.c:3133 Auto Changing port from 10.3.212.105:8024 to 10.3.212.105:12016 Thank you, A.C. CC: freeswitch-users@lists.freeswitch.org From: steveayre@gmail.com Date: Fri, 13 Jul 2012 13:30:38 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Can you supply a pastebin of adebug log of the call? Steve on iPhone On 12 Jul 2012, at 21:39, Adelia C. <hexade@hotmail.com> wrote: We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. Call: Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. RTP streams, as negotiated (SDP): Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) RTP streams, as sent/receiced: FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] All this from Wireshark trace. I tried the following, with no success: 1. Added to MyFSApp.xml and dialplan\public.xml: <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Restarted, FS, same result. 2. Commented out <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="use-rtp-timer" value="true"/> Restarted, FS, same result. 3. Started FS with -nonat, then nonatmap Same result. Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old. What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? Thanks, A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Guys, any idea what this is not working? - http://pastebin.freeswitch.org/19510 (logs from call). FS is enforcing symmetrical RTP in a config with an asymmetrical-only B2BUA. It starts by sending RTP to the negotiated port (remote destination) but switches to the actual remote source in ~.3-.5s.I tried setting disable_rtp_auto_adjust=true. Thanks,A.C. From: hexade@hotmail.com To: freeswitch-users@lists.freeswitch.org Date: Fri, 13 Jul 2012 13:37:25 -0400 Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Thank you Steve - here it is - http://pastebin.freeswitch.org/19510. Fresh call, new ports. Notice this line, this is what I want to stop: 2012-07-13 10:18:42.935692 [INFO] switch_rtp.c:3133 Auto Changing port from 10.3.212.105:8024 to 10.3.212.105:12016 Thank you, A.C. CC: freeswitch-users@lists.freeswitch.org From: steveayre@gmail.com Date: Fri, 13 Jul 2012 13:30:38 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Can you supply a pastebin of adebug log of the call? Steve on iPhone
participants (2)
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Adelia C.
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Steven Ayre