Issue with SIP audio calls in SIP as RTMP .
- Thread starter Karan Bhansali
- Start date
Muhammad Rizwan
Member
this is a very major issue if DTMF does not work. please give me estimate time on this fix.
Hello
WCS supports two DTMF modes:
Third mode "DTMF Inband" is currently unsupported in WCS server.
So to get this working you have to either migrate to another PBX or setup DTMF over RFC2833 / SIP INFO on your PBX.
This feature is currently not supported in WCS server.
Please read more about DTMF types:
https://www.voip-info.org/sip-dtmf-signalling/
WCS supports two DTMF modes:
- DTMF over RFC2833
- DTMF over SIP INFO
Third mode "DTMF Inband" is currently unsupported in WCS server.
So to get this working you have to either migrate to another PBX or setup DTMF over RFC2833 / SIP INFO on your PBX.
eyebeam uses "DTMF Inband"
This feature is currently not supported in WCS server.
Please read more about DTMF types:
https://www.voip-info.org/sip-dtmf-signalling/
Muhammad Rizwan
Member
well. i last version Web Call Server 5.0 every thing was working fine on same pbx dtmf was working voice was working but suddenly out no where voice stopped working on Web Call Server 5.0? why why why???
now you are saying on upgraded version Web Call Server 5.2 DTMF is not working due to pbx support. my sip server support type of DTMF. even these given below.
now you are saying on upgraded version Web Call Server 5.2 DTMF is not working due to pbx support. my sip server support type of DTMF. even these given below.
- DTMF over RFC2833
- DTMF over SIP INFO
Muhammad Rizwan
Member
i have even checked DTMF with asterisk from eyebeam softphone that is working fine but then i call from webrtc phone-ui it does not work....... please please please fix this issues..............................................
Muhammad Rizwan
Member
now i have tried calling using Web Call Server 5.0 now DTMF is working. please find given below screenshot. now on this version there is voice issue. there is no voice. just fix this voice issue i would like to use this Web Call Server 5.0 version
http://prntscr.com/mz1mr8
please fix this voice issue... this is very urgently required. Web Call Server server is not working properly since i did the upgrade.....
http://prntscr.com/mz1mr8
please fix this voice issue... this is very urgently required. Web Call Server server is not working properly since i did the upgrade.....
Your SIP gateway (PBX) answers in SDP
As you can see, your PBX does not indicate telephone-event (DTMF).
And therefore your PBX does not work properly. Not WCS.
You claim WCS 5.0 sends DTMF properly. Well this is a bug in WCS 5.0, because WCS should NOT send DTMF if DTMF (telephone-event) is not presented in 200 OK SDP.
Please check out RFC4317
https://tools.ietf.org/html/rfc4317
As you can see, PBX MUST answer with telephone-event to indicate it supports DTMF.
Your PBX does not indicate the telephone-event. So WCS does not send DTMF and it is correct. No bug here.
Therefore no bug on our end and you have to fix your PBX to get this working.
Well. We can try to find a workaround and consider a tweak allowing to get this working with your PBX. We will inform you once we have any news on this way.
But for now, primary issue is your PBX that does not work according SIP specifications and you have to contact PBX support or migrate to another PBX fix the issue.
If you have any objections, please provide a proof that your PBX supports RFC2833 or SIP INFO and works properly according SIP specifications.
Code:
SIP/2.0 200 OK
m=audio 8376 RTP/AVP 18
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=sendrecvAnd therefore your PBX does not work properly. Not WCS.
You claim WCS 5.0 sends DTMF properly. Well this is a bug in WCS 5.0, because WCS should NOT send DTMF if DTMF (telephone-event) is not presented in 200 OK SDP.
Please check out RFC4317
https://tools.ietf.org/html/rfc4317
As you can see, PBX MUST answer with telephone-event to indicate it supports DTMF.
Code:
[Answer]
v=0
o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
s=
c=IN IP4 host.biloxi.example.com
t=0 0
m=audio 49172 RTP/AVP 97
a=rtpmap:97 iLBC/8000
m=audio 49174 RTP/AVP 98
a=rtpmap:98 telephone-event/8000
a=recvonlyTherefore no bug on our end and you have to fix your PBX to get this working.
Well. We can try to find a workaround and consider a tweak allowing to get this working with your PBX. We will inform you once we have any news on this way.
But for now, primary issue is your PBX that does not work according SIP specifications and you have to contact PBX support or migrate to another PBX fix the issue.
If you have any objections, please provide a proof that your PBX supports RFC2833 or SIP INFO and works properly according SIP specifications.
Muhammad Rizwan
Member
i have objections,
i have 4 different switches Sippy, VOS3000, VoipSwitch & Asterisk WCS 5.1 and 5.2 both have issues with DTMF. this is not possible that all 4 PBX have issues where as your WCS 5.1 and 5.2 does not have any bugs.. anyway please find given below screenshot where i have personally set DTMF = RFC2833
http://prntscr.com/mz5yew
using same account i made call from WCS 5.0 and DTMF worked fine but using WCS 5.1 & 5.2 it didn't work.
ok look easy way around even if you say it was a bug in WCS 5.0 that WCS should NOT send DTMF if DTMF (telephone-event) is not presented in 200 OK SDP. then make it same in in WCS 5.2 .. alteast it works with DTMF even its a bug.......
Look i am really tired from all these issues please help me to fix it ASAP.
i have been using flashphoner from a very long time since you had flash version. i always recommend flashphoner to others when it comes to WebRTC to SIP .. but trust me with these many issues i am losing my trust..
Thanks
i have 4 different switches Sippy, VOS3000, VoipSwitch & Asterisk WCS 5.1 and 5.2 both have issues with DTMF. this is not possible that all 4 PBX have issues where as your WCS 5.1 and 5.2 does not have any bugs.. anyway please find given below screenshot where i have personally set DTMF = RFC2833
http://prntscr.com/mz5yew
using same account i made call from WCS 5.0 and DTMF worked fine but using WCS 5.1 & 5.2 it didn't work.
ok look easy way around even if you say it was a bug in WCS 5.0 that WCS should NOT send DTMF if DTMF (telephone-event) is not presented in 200 OK SDP. then make it same in in WCS 5.2 .. alteast it works with DTMF even its a bug.......
Look i am really tired from all these issues please help me to fix it ASAP.
i have been using flashphoner from a very long time since you had flash version. i always recommend flashphoner to others when it comes to WebRTC to SIP .. but trust me with these many issues i am losing my trust..
Thanks
Hello.
We have tested the latest build 5.2.36 with OpenSIPS and Asterisk, with WCS default settings. WCS sends RFC2833 DTMF correctly, see traffic dump analisys screenshot
So please update to this build and provide us traffic dumps for all of your PBXes where WCS does not work:
We have tested the latest build 5.2.36 with OpenSIPS and Asterisk, with WCS default settings. WCS sends RFC2833 DTMF correctly, see traffic dump analisys screenshot
So please update to this build and provide us traffic dumps for all of your PBXes where WCS does not work:
Also, provide us SIP accounts on those PBXes for testing.
Muhammad Rizwan
Member
i have send you email with account details please check it..
Thanks
Thanks
Hello.
We'll make a fix that should force DTMF sending even if no telephone-event codec in SDP (internal ticket WCS-1860).
Also it seems like the problem is in WCS codecs configuration. In flashphoner.properties you have sent default codecs settings are used:
So WCS set to INVITE SDP the following codecs:
plus additional codecs from browser.
As a result, INVITE SDP packet size exceeds MTU limit (that is 1500 bytes), and SDP is sent by two packets. The most of PBX does not allow SDP fragmentation and just drop second packet, so some codecs becomes invisible to PBX. Therefore WCS can register on SIP switch, but can not establish a call (for example with 408 Request timeout message in logs), or can not send DTMF even if telephone-event is supported by SIP switch.
A workaround is to exclude some codecs from SIP call and exclude additional browsers codecs as described here (point 4). For example, to make SIP calls using only G729 for audio, VP8 for video and telephone-event for DTMF you should set up WCS codecs as follows
We'll make a fix that should force DTMF sending even if no telephone-event codec in SDP (internal ticket WCS-1860).
Also it seems like the problem is in WCS codecs configuration. In flashphoner.properties you have sent default codecs settings are used:
Code:
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip=mpeg4-generic,flv,mpv
Code:
opus,alaw,ulaw,g729,speex16,g722,telephone-event,h264,vp8As a result, INVITE SDP packet size exceeds MTU limit (that is 1500 bytes), and SDP is sent by two packets. The most of PBX does not allow SDP fragmentation and just drop second packet, so some codecs becomes invisible to PBX. Therefore WCS can register on SIP switch, but can not establish a call (for example with 408 Request timeout message in logs), or can not send DTMF even if telephone-event is supported by SIP switch.
A workaround is to exclude some codecs from SIP call and exclude additional browsers codecs as described here (point 4). For example, to make SIP calls using only G729 for audio, VP8 for video and telephone-event for DTMF you should set up WCS codecs as follows
Code:
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip=mpeg4-generic,flv,mpv,opus,alaw,ulaw,speex16,g722,h264
allow_outside_codecs=falseMuhammad Rizwan
Member
hello,
i have set given below settings and now i tried to make call and there is no sound once the call is connected. whereas on asterisk DTMF is working with this setting.
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip=mpeg4-generic,flv,mpv,opus,alaw,ulaw,speex16,g722,h264
allow_outside_codecs=false
i have set given below settings and now i tried to make call and there is no sound once the call is connected. whereas on asterisk DTMF is working with this setting.
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip=mpeg4-generic,flv,mpv,opus,alaw,ulaw,speex16,g722,h264
allow_outside_codecs=false
In example above, we enabled G729 codec only. Try to enable more audio codecs. Also, if you do not use video calls at all, you can disable vp8 codec.
Also, you can enable TCP usage for SIP signaling with this setting
Code:
sip_force_tcp=true
Last edited:
Muhammad Rizwan
Member
hello,
i have even set my SIP server on TCP port but still no sound please check this image http://prntscr.com/n0ebr8 i tried on different SIp port but result was same. further more Sippy does not support TCP SIP commincation so voice should also has to be fixed using UDP.
also i am sending my flashphoner.properties config..
#webrtc ports range
media_port_from =31001
media_port_to =32000
#codecs
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv,opus,alaw,ulaw,speex16,g722,h264
allow_outside_codecs =false
sip_force_tcp =true
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,mpv
#websocket ports
ws.port =8080
wss.port =8443
please check and let me know..
i have even set my SIP server on TCP port but still no sound please check this image http://prntscr.com/n0ebr8 i tried on different SIp port but result was same. further more Sippy does not support TCP SIP commincation so voice should also has to be fixed using UDP.
also i am sending my flashphoner.properties config..
#webrtc ports range
media_port_from =31001
media_port_to =32000
#codecs
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv,opus,alaw,ulaw,speex16,g722,h264
allow_outside_codecs =false
sip_force_tcp =true
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,mpv
#websocket ports
ws.port =8080
wss.port =8443
please check and let me know..
You left only g729 codec for SIP calls, this codec does not supported by Chrome browser, so there's no sound. Choose another audio codec: opus, alaw, ulaw.
to disable TCP SIP support. Also, set this parameter
and check if DTMF work with Sippy.
Then comment the setting
Code:
sip_force_tcp=true
Code:
dtmf=INFOMuhammad Rizwan
Member
i am really tired of this DTMF issues now. i installed FlashphonerWebCallServer-5.2.4.tar.gz this version and in flashphoner.properties i set given below configs and DTMF started working fine on my all switches sippy, voipswitch, asterisk. now just after few days it stopped working again. why is this happing? is it some sort of magic? it started working and stopped working again. why flashphoner is not stayble like before? please please fix this issue............ i am really tired now
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =ulaw,alaw,g729,g722,mpeg4-generic,vp8,mpv
allow_outside_codecs =false
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =ulaw,alaw,g729,g722,mpeg4-generic,vp8,mpv
allow_outside_codecs =false
Hello.
1. Adjust codecs configuration (this post)
2. Set up your SIP switches to use TCP transport and set WCS to use it too (this post)
3. A possible trick for Sippy only (this post)
No, it's a kind of network issue. It looks like MTU size on some network hardware (switch, router etc) changes, this leads to SIP SDP fragmentation. In this thread we described all the possible ways to prevent it:
1. Adjust codecs configuration (this post)
2. Set up your SIP switches to use TCP transport and set WCS to use it too (this post)
3. A possible trick for Sippy only (this post)
Muhammad Rizwan
Member
hi,
the setting you sent me i have already tested them before but it didn't work but with new release it started working but automatically it got stopped again.. please connect to my flashphoner server and fix this issue.
Thanks
the setting you sent me i have already tested them before but it didn't work but with new release it started working but automatically it got stopped again.. please connect to my flashphoner server and fix this issue.
Thanks
Hello.
Please send SSH credentials to support@flashphoner.com. We will check.
Please send SSH credentials to support@flashphoner.com. We will check.
Muhammad Rizwan
Member
hello team,
Details has been sent to your email. thanks
Details has been sent to your email. thanks