13 Oct
2017
13 Oct
'17
9:58 p.m.
Hi all!
Some time ago Chromium browser sets rtcpMuxPolicy: required by default
(soon on Chrome 58)
It means that webRTC based clients not accepts
a=rtcp:31757
And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC
client (SIP client is initiator of call)
When sip client sends invite it has
a=rtcp:33445
Means it wants 2 different prots for RTCP and RTP
As expected for this case webRTC client says 488 Not accessible here
instead of 200 resonse
I suppose rtpengine module should hept to handle it but i can not find any
key how to do it
I added form rtpengine_manage()
rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux"
But not removes a=rtcp and ice cadidate prepeared for it.
Suppose removing a=rtcp:12345 will gives just an issue for RTP session.
Does rtpengine module have some keys for resole this issue?
13 Oct
13 Oct
11:12 p.m.
Sorry:
small fix
webRTC clients accepts
a=rtcp:<port>
but port suppose should be same with
m=audio
2017-10-13 22:58 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
Hi all!
Some time ago Chromium browser sets rtcpMuxPolicy: required by default
(soon on Chrome 58)
It means that webRTC based clients not accepts
a=rtcp:31757
And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC
client (SIP client is initiator of call)
When sip client sends invite it has
a=rtcp:33445
Means it wants 2 different prots for RTCP and RTP
As expected for this case webRTC client says 488 Not accessible here
instead of 200 resonse
I suppose rtpengine module should hept to handle it but i can not find any
key how to do it
I added form rtpengine_manage()
rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux"
But not removes a=rtcp and ice cadidate prepeared for it.
Suppose removing a=rtcp:12345 will gives just an issue for RTP session.
Does rtpengine module have some keys for resole this issue?
14 Oct
14 Oct
10 a.m.
Please look one more example o SDP with rtcp-mux
------------------------------------------------------------------------
INVITE sip:3000@pbx.rcsnet.ru:5060 SIP/2.0
Via: SIP/2.0/UDP 91.103.196.12;rport;branch=z9hG4bK32SrX56KHZgya
Max-Forwards: 70
From: "" <sip:0000000000@91.103.196.12>;tag=3Z0mjUyp1matN
To: <sip:3000@pbx.rcsnet.ru:5060>
Call-ID: e3be7793-cd5a-1235-d195-005056be15c6
CSeq: 108480740 INVITE
Contact: <sip:mod_sofia@91.103.196.12:5060>
User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20170615T144716Z~5f5fb33ea9~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 670
X-FS-Support: update_display,send_info
Remote-Party-ID:
<sip:0000000000@91.103.196.12>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1497607971 1497607972 IN IP4 91.103.196.12
s=FreeSWITCH
c=IN IP4 91.103.196.12
t=0 0
m=audio 25638 RTP/AVP 102 9 0 8 104 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000;
maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 telephone-event/48000
a=fmtp:104 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-mux
a=rtcp:25638 IN IP4 91.103.196.12
a=ptime:20
m=video 23108 RTP/AVP 103
b=AS:1024
a=rtpmap:103 VP8/90000
a=rtcp-fb:103 ccm fir
a=rtcp-fb:103 ccm tmmbr
a=rtcp-fb:103 nack
a=rtcp-fb:103 nack pli
------------------------------------------------------------------------
Log https://freeswitch.org/jira/secure/attachment/26623/originate_log.txt
Ticket https://freeswitch.org/jira/browse/FS-10400
Sergey
сб, 14 окт. 2017 г. в 0:13, Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
Sorry:
small fix
webRTC clients accepts
a=rtcp:<port>
but port suppose should be same with
m=audio
2017-10-13 22:58 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
Hi all!
Some time ago Chromium browser sets rtcpMuxPolicy: required by default
(soon on Chrome 58)
It means that webRTC based clients not accepts
a=rtcp:31757
And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC
client (SIP client is initiator of call)
When sip client sends invite it has
a=rtcp:33445
Means it wants 2 different prots for RTCP and RTP
As expected for this case webRTC client says 488 Not accessible here
instead of 200 resonse
I suppose rtpengine module should hept to handle it but i can not find
any key how to do it
I added form rtpengine_manage()
rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux"
But not removes a=rtcp and ice cadidate prepeared for it.
Suppose removing a=rtcp:12345 will gives just an issue for RTP session.
Does rtpengine module have some keys for resole this issue?
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
10:24 a.m.
Looks like here is FreeSWITH fixes it from it's side with rtsp-mux=true.
As i said above - a=rtcp:<port> should present but should use same port
with m=audio.
I know that freeswith allows rtcp-mux, but asterisk, for example does not.
Thats why im asking how to handle this issue with rtpengine if it possible.
On Oct 14, 2017 11:02 AM, "Sergey Safarov" <s.safarov(a)gmail.com> wrote:
Please look one more example o SDP with rtcp-mux
------------------------------------------------------------------------
INVITE sip:3000@pbx.rcsnet.ru:5060 SIP/2.0
Via: SIP/2.0/UDP 91.103.196.12;rport;branch=z9hG4bK32SrX56KHZgya
Max-Forwards: 70
From: "" <sip:0000000000@91.103.196.12>;tag=3Z0mjUyp1matN
To: <sip:3000@pbx.rcsnet.ru:5060>
Call-ID: e3be7793-cd5a-1235-d195-005056be15c6
CSeq: 108480740 INVITE
Contact: <sip:mod_sofia@91.103.196.12:5060>
User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20170615T144716Z~5f5fb33ea9~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER,
NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 670
X-FS-Support: update_display,send_info
Remote-Party-ID:
<sip:0000000000@91.103.196.12>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1497607971 1497607972 IN IP4 91.103.196.12
s=FreeSWITCH
c=IN IP4 91.103.196.12
t=0 0
m=audio 25638 RTP/AVP 102 9 0 8 104 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20;
minptime=10; maxptime=40
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 telephone-event/48000
a=fmtp:104 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-mux
a=rtcp:25638 IN IP4 91.103.196.12
a=ptime:20
m=video 23108 RTP/AVP 103
b=AS:1024
a=rtpmap:103 VP8/90000
a=rtcp-fb:103 ccm fir
a=rtcp-fb:103 ccm tmmbr
a=rtcp-fb:103 nack
a=rtcp-fb:103 nack pli
------------------------------------------------------------------------
Log https://freeswitch.org/jira/secure/attachment/26623/originate_log.txt
Ticket https://freeswitch.org/jira/browse/FS-10400
Sergey
сб, 14 окт. 2017 г. в 0:13, Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
Sorry:
small fix
webRTC clients accepts
a=rtcp:<port>
but port suppose should be same with
m=audio
2017-10-13 22:58 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi all!
Some time ago Chromium browser sets rtcpMuxPolicy: required by default
(soon on Chrome 58)
It means that webRTC based clients not accepts
a=rtcp:31757
And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC
client (SIP client is initiator of call)
When sip client sends invite it has
a=rtcp:33445
Means it wants 2 different prots for RTCP and RTP
As expected for this case webRTC client says 488 Not accessible here
instead of 200 resonse
I suppose rtpengine module should hept to handle it but i can not find
any key how to do it
I added form rtpengine_manage()
rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux"
But not removes a=rtcp and ice cadidate prepeared for it.
Suppose removing a=rtcp:12345 will gives just an issue for RTP session.
Does rtpengine module have some keys for resole this issue?
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
2410
days inactive
2411
days old
3 comments
2 participants
participants (2)
-
Sergey Safarov -
Yuriy Gorlichenko