18 Dec
2014
18 Dec
'14
6:11 p.m.
Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only with
Firefox browser), even with video. But in other directions i have some
issues because of lack of RTP delivery or RTP timeouts.
I have some logs to show you regarding this:
https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
work).
I would really love to get some quick help from anyone. For direct
manual fixing, I can give a couple of hundreds of bucks.
Looking forward impatiently for reply from anyone having something to say.
--
Andrey Utkin
18 Dec
18 Dec
6:30 p.m.
On 12/18/14 12:11, Andrey Utkin wrote:
Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only with
Firefox browser), even with video. But in other directions i have some
issues because of lack of RTP delivery or RTP timeouts.
I have some logs to show you regarding this:
https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
work).
I would really love to get some quick help from anyone. For direct
manual fixing, I can give a couple of hundreds of bucks.
Looking forward impatiently for reply from anyone having something to say.
Write error on RTP socket usually indicates an incorrect network setup,
for example trying to use a source address for IP packets which isn't
bound to any local network interface (especially if you're sitting
behind NAT), or local iptables rules rejecting outgoing IP packets, or
missing IP routes to the destination. Perhaps post your network setup
and the CLI arguments to rtpengine you're using.
cheers
6:55 p.m.
2014-12-18 19:30 GMT+02:00 Richard Fuchs <rfuchs(a)sipwise.com>om>:
Write error on RTP socket usually indicates an
incorrect network setup,
for example trying to use a source address for IP packets which isn't
bound to any local network interface (especially if you're sitting
behind NAT), or local iptables rules rejecting outgoing IP packets, or
missing IP routes to the destination. Perhaps post your network setup
and the CLI arguments to rtpengine you're using.
I use VPS from DigitalOcean. Likely no NAT. Only one public IP, and
one domain name used (whdd.org). Exactly the same situation was with
Amazon VPS behind amazonish NAT.
My configs are here: https://github.com/krieger-od/webrtc_kamailio_etc
(only kamailio.cfg and modules.cfg are used, kamailio*.cfg are not).
Also there you can check command line used to run rtpengine. To put it
there, it is
/home/krieger/rtpengine/daemon/rtpengine --interface=188.226.241.236
--listen-ng=127.0.0.1:22222 -m 30000 -M 35000 --foreground
--log-stderr
--
Andrey Utkin
7:05 p.m.
On 12/18/14 12:55, Andrey Utkin wrote:
2014-12-18 19:30 GMT+02:00 Richard Fuchs
<rfuchs(a)sipwise.com>om>:
Amazon NAT is exactly why I've mentioned it, because on an Amazon
system, if you don't use the --interface option correctly
($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
that show in your log.
But these configs and this CLI line don't match the logs you posted
earlier. Is it also write errors you're getting with those?
cheers
Write error on RTP socket usually indicates an
incorrect network setup,
for example trying to use a source address for IP packets which isn't
bound to any local network interface (especially if you're sitting
behind NAT), or local iptables rules rejecting outgoing IP packets, or
missing IP routes to the destination. Perhaps post your network setup
and the CLI arguments to rtpengine you're using.
I use VPS from DigitalOcean. Likely no NAT. Only one public IP, and
one domain name used (whdd.org). Exactly the same situation was with
Amazon VPS behind amazonish NAT.
My configs are here: https://github.com/krieger-od/webrtc_kamailio_etc
(only kamailio.cfg and modules.cfg are used, kamailio*.cfg are not).
Also there you can check command line used to run rtpengine. To put it
there, it is
/home/krieger/rtpengine/daemon/rtpengine --interface=188.226.241.236
--listen-ng=127.0.0.1:22222 -m 30000 -M 35000 --foreground
--log-stderr
7:09 p.m.
2014-12-18 20:05 GMT+02:00 Richard Fuchs <rfuchs(a)sipwise.com>om>:
Amazon NAT is exactly why I've mentioned it,
because on an Amazon
system, if you don't use the --interface option correctly
($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
that show in your log.
Thank you, will try with such notation on amazon host now.
But these configs and this CLI line don't match
the logs you posted
earlier. Is it also write errors you're getting with those?
Will provide you shortly with logs from DigitalOcean VPS instance,
configs for which i've published on github.
--
Andrey Utkin
7:16 p.m.
2014-12-18 20:09 GMT+02:00 Andrey Utkin <andrey.krieger.utkin(a)gmail.com>om>:
2014-12-18 20:05 GMT+02:00 Richard Fuchs
<rfuchs(a)sipwise.com>om>:
This alone haven't helped with Amazon instance.
Will make tests from DigitalOcean instance and provide you with logs.
--
Andrey Utkin
Amazon NAT is exactly why I've mentioned it,
because on an Amazon
system, if you don't use the --interface option correctly
($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
that show in your log.
Thank you, will try with such notation on amazon host now.
7:38 p.m.
This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
This doesn't work: few seconds after answer, there's no media from
remote subscriber, then linphone android agent hangs up; sipml hangs
pretending being in call.
rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde
--
Andrey Utkin
7:51 p.m.
2014-12-18 20:38 GMT+02:00 Andrey Utkin <andrey.krieger.utkin(a)gmail.com>om>:
This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
This doesn't work: few seconds after answer, there's no media from
remote subscriber, then linphone android agent hangs up; sipml hangs
pretending being in call.
rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde
Doesn't work in different way:
jssip web agent calls to android linphone: call is connected, but
android doesn't see peer's video, and jssip doesn't have peer's audio:
rtpengine: https://gist.github.com/krieger-od/cd63ef99d06212b30379
kamailio: https://gist.github.com/krieger-od/55c4b5785c5821315955
ngrep: https://gist.github.com/krieger-od/650575c0f96285d78d17
--
Andrey Utkin
8:24 p.m.
Also I encounter such issue: even in the working scenario, the hangup
of one peer doesn't make the call end for another peer.
rtpengine: https://gist.github.com/krieger-od/1cfe84b53dc0d29cfb90
kamailio: https://gist.github.com/krieger-od/11c6bbf7dad15382e81b
ngrep: https://gist.github.com/krieger-od/ec6c21c6f9dde1400578
Also, this deploy of http://vmwrtc.intersog.com/sipml5/call.htm works
ok, and official one http://sipml5.org/call.htm?svn=222 has more
issues.
--
Andrey Utkin
8:28 p.m.
On 12/18/14 13:38, Andrey Utkin wrote:
This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
This doesn't work: few seconds after answer, there's no media from
remote subscriber, then linphone android agent hangs up; sipml hangs
pretending being in call.
rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde
Couple of things to try here...
In the second case, Firefox kept sending answers (200 OK) for no good
reason. Check Firefox's debug log / JS console / whatever if there's any
error or warnings messages.
Try running rtpengine with the --dtls-passive switch, even though that
shouldn't make a difference after ICE went through (which it does).
Add rtcp-mux=demux to your rtpengine flags in your Kamailio config,
should be safe to use everywhere.
It's curious that Firefox was using RTP/SAVPF as protocol in one call
and UDP/TLS/RTP/SAVPF in another. I don't suspect that this would cause
a problem, but it's worth trying to specify the protocol not as
RTP/SAVPF but rather as transport-protocol=UDP/TLS/RTP/SAVPF. Note that
in the case of answers, you don't need to specify any protocol at all.
In the third case, the audio stream was setup as a=sendonly, explaining
the one-way audio. Probably caused by Firefox not being able to access
the playback device.
Last but not least, try a different browser, either a newer version of
Firefox (it's up to v34 I see), or Chrome.
As for the call teardown, it's Kamailio's responsibility (through the
config) to let rtpengine know that the call has been terminated, for
example when handling the BYE method. Otherwise you'll see timeouts in
rtpengine. Even though the BYE should still be relayed to the other SIP
end, causing the call to be hung up there.
cheers
19 Dec
19 Dec
8:44 a.m.
Richard Fuchs writes:
In the third case, the audio stream was setup as
a=sendonly, explaining
the one-way audio. Probably caused by Firefox not being able to access
the playback device.
as i reported in another message, i have also lately noticed in my tests
that firefox sends invite with a=sendonly attribute. still don't know
why, because the same firefox can play audio without problems using the
html <audio> tag.
-- juha
3436
days inactive
3437
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10 comments
3 participants
participants (3)
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Andrey Utkin -
Juha Heinanen -
Richard Fuchs