Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP) FreeSWITCH after answering the call says 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0* and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back? On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same. FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97 Appreciate some input to figure out this problem. Thanks, Naseer
PCMU:0 is normal. 0 is the payload type for PCMU. Perhaps you have the firewall enabled on the machines themselves ? Flush your iptables rules, just to see if it works. On Thu, Jul 9, 2020 at 8:17 PM Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
There is No firewall. Already turned off for the sake of testing. on the Same machine just turn off FreeSWITCH and started Asterisk, able to hear audio both ways. Switch back to FreeSWITCH and get one way only. When capturing RTP I can hear far (Play audio stream in Wireshark) but FreeSWITCH won’t try to send any RTPs out. From: Dragos Oancea <dragos@freeswitch.org> <dragos@freeswitch.org> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 9, 2020 at 22:55:22 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC PCMU:0 is normal. 0 is the payload type for PCMU.
Perhaps you have the firewall enabled on the machines themselves ? Flush your iptables rules, just to see if it works.
On Thu, Jul 9, 2020 at 8:17 PM Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction. I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix. Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur. Sent from my Windows 10 device From: Muhammad Naseer Bhatti<mailto:nbhatti@gmail.com> Sent: 09 July 2020 18:44 To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP) FreeSWITCH after answering the call says 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130<mailto:04232152273@43.225.99.130> PCMU/8000 40 ms 320 samples 64000 bits 1 channels 2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130<mailto:04232152273@43.225.99.130> Original read codec set to PCMU:0 and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back? On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same. FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97 Appreciate some input to figure out this problem. Thanks, Naseer
You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue. On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer@outlook.com> wrote:
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.
I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.
Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.
Sent from my Windows 10 device
*From: *Muhammad Naseer Bhatti <nbhatti@gmail.com> *Sent: *09 July 2020 18:44 *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
Tried ptime 20 and 40 both on FreeSWITCH. The behavior is the same with both. I’ll have not changed in the client, will see if there is even an option, but this is strange behavior never seen that before. From: Dragos Oancea <dragos@freeswitch.org> <dragos@freeswitch.org> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 10, 2020 at 9:36:47 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue. On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer@outlook.com> wrote:
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.
I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.
Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.
Sent from my Windows 10 device
*From: *Muhammad Naseer Bhatti <nbhatti@gmail.com> *Sent: *09 July 2020 18:44 *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com 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 https://freeswitch.com
Can you share a complete sip trace? And fs Sofia config? On Fri, 10 Jul 2020 at 19:30, Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
Tried ptime 20 and 40 both on FreeSWITCH. The behavior is the same with both. I’ll have not changed in the client, will see if there is even an option, but this is strange behavior never seen that before.
From: Dragos Oancea <dragos@freeswitch.org> <dragos@freeswitch.org> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 10, 2020 at 9:36:47 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue.
On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer@outlook.com> wrote:
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.
I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.
Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.
Sent from my Windows 10 device
*From: *Muhammad Naseer Bhatti <nbhatti@gmail.com> *Sent: *09 July 2020 18:44 *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
SIP trace is at https://pastebin.freeswitch.org/view/0925119c and log at https://pastebin.freeswitch.org/view/69b5f68c and sofia config is at https://pastebin.freeswitch.org/view/77c979ef I stripeed down the comments and generated this from freeswitch.xml.fsxml to ensure this shows only the applied config. From: David Villasmil <david.villasmil.work@gmail.com> <david.villasmil.work@gmail.com> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 11, 2020 at 7:58:29 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC Can you share a complete sip trace? And fs Sofia config? On Fri, 10 Jul 2020 at 19:30, Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
Tried ptime 20 and 40 both on FreeSWITCH. The behavior is the same with both. I’ll have not changed in the client, will see if there is even an option, but this is strange behavior never seen that before.
From: Dragos Oancea <dragos@freeswitch.org> <dragos@freeswitch.org> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 10, 2020 at 9:36:47 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue.
On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer@outlook.com> wrote:
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.
I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.
Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.
Sent from my Windows 10 device
*From: *Muhammad Naseer Bhatti <nbhatti@gmail.com> *Sent: *09 July 2020 18:44 *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com 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 https://freeswitch.com
There’s only one leg in the trace, we’d need both legs. Also, try not to change the IPs, since that might be the problem. But you can check this yourself: A (ip x.x.x.1): INVITE B With IP x.x.x.1 in “c=“ and “o=“ in the SDP B (x.x.x.2) 200 OK With IP x.x.x.2 in “c=“ and “o=“ in the SDP Same goes for B->C, B must sent ITS OWN IP to C and C must offer its own IP. Also, how are you starting freeswitch? On Sat, 11 Jul 2020 at 09:50, Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
SIP trace is at https://pastebin.freeswitch.org/view/0925119c and log at https://pastebin.freeswitch.org/view/69b5f68c and sofia config is at https://pastebin.freeswitch.org/view/77c979ef I stripeed down the comments and generated this from freeswitch.xml.fsxml to ensure this shows only the applied config.
From: David Villasmil <david.villasmil.work@gmail.com> <david.villasmil.work@gmail.com> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 11, 2020 at 7:58:29 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Can you share a complete sip trace? And fs Sofia config?
On Fri, 10 Jul 2020 at 19:30, Muhammad Naseer Bhatti <nbhatti@gmail.com> wrote:
Tried ptime 20 and 40 both on FreeSWITCH. The behavior is the same with both. I’ll have not changed in the client, will see if there is even an option, but this is strange behavior never seen that before.
From: Dragos Oancea <dragos@freeswitch.org> <dragos@freeswitch.org> Reply: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Date: July 10, 2020 at 9:36:47 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
You have ptime 40, does it work the common way (with ptime 20) ? Change it in the client, your client makes an offer with ptime 40. It should not be a problem (40 ms is ok too) , but maybe we get a hint on this one way audio issue.
On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer@outlook.com> wrote:
Don't know if it's the same problem but I incorporated FreeSWITCH code into a product a couple of years ago (master code was marked as 1.9.x) and whilst the (FreeSWITCH) server worked fine, a client built with that version of the software would only give audio in one direction.
I figured out a fix (to switch_ivr_originate.c) and raised a Jira but didn't provide the fix as I wasn’t sure the change I had made was the best way to resolve the issue. I assumed that someone with a better understanding of the code would be better placed to determine the most appropriate fix.
Interestingly if I used a much older version of the FreeSWITCH software in the client (like that shipping with the FSClient software) the problem did not occur.
Sent from my Windows 10 device
*From: *Muhammad Naseer Bhatti <nbhatti@gmail.com> *Sent: *09 July 2020 18:44 *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is
SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
FreeSWITCH after answering the call says
2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273@43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273@43.225.99.130 <04232152273@43.225.99.130> Original read codec set to PCMU:0*
and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back?
On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same.
FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
Appreciate some input to figure out this problem.
Thanks, Naseer _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
--
Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
participants (4)
-
David Villasmil
-
Dragos Oancea
-
Muhammad Naseer Bhatti
-
Paul Mateer