[sip-comm] Tried sound and video: Nothing worked


#1

Hello,

I tried to establish some voice (and possibly video) calls using today's nightly build. Unfortunately, neither voice nor video worked.

I tried both Jabber and SIP.

Using SIP, the client was able to register with my SIP provider and show incoming calls. When I tried to answer the calls, there was no audio and the call died after a few seconds. I was unable to find out how to call IP addresses directly, with no intermediate registrars and proxies. (Something like that is simply necessary and supported by Linphone or Twinkle, for example.) (I also did not find any possibility to use IPv6.) Outgoing calls died in the moment when the other party tried to answer them.

Three other SIP softphones and one hardware phone worked fine under exactly the same conditions. (Linphone could even establish a video call using both IPv6 and (tunneled) IPv4.)

Then I enabled a media proxy on my Jabber server and tried to establish calls via Jabber. (I use the OpenFire server. Don't know whether its media proxy is compatible or necessary, but switched it on anyway.) I disabled the Jaber server's firewall completely. Unfortunately, calls via Jabber behaved the same way as calls via SIP. Incoming calls died after a few seconds when answered. There was no audio. Outgoing calls showed Alerting the user and played a ringtone. From the caller's point of view, they died immdiately when the other party accepted them.

Well,this was somewhat frustrating... Obviously, video remained a pipe dream when even sound did not work. It seems that Jabber and SIP work fine, but the RTP stream gets stuck somewhere. :frowning:

It would be great if I could establish direct IP-to-IP calls within LAN (or IPv6). The problem would become much easier to diagnose. Maybe I just misconfigured or misunderstood something.

Regards,

Andrej

···

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#2

Hello Andrej,

Thanks for reporting your experience here.

--more inline

Andrej Podzimek wrote:

Hello,

I tried to establish some voice (and possibly video) calls using
today's nightly build. Unfortunately, neither voice nor video worked.

Sorry to hear that.

I tried both Jabber and SIP.

As mentioned on a post on our dev list today, support for XMPP/Jingle is
in an experimental state. We'll probably allocate some effort there in
the following months but it's state is quite shaky now as you have noticed.

Using SIP, the client was able to register with my SIP provider and
show incoming calls. When I tried to answer the calls, there was no
audio and the call died after a few seconds.

I see. I am afraid however that there's little we could do to help with
"didn't work" reports. In order to know what really happened we'd need
to have a look at the log files for a failed session. A
tcpdump/wireshark capture would also help if it's not too much trouble.
You may want to have a look at:

http://www.sip-communicator.org/index.php/Documentation/ReportingBugs

I was unable to find out how to call IP addresses directly,

Simply dialing username@<ip-address> or even <ip-address> should work.
If it didn't, then the problem was probably caused by something else and
not support of ip-address dialing itself.

with no intermediate registrars and proxies.

You can create registrarless accounts the same way you create regular
ones. Only, instead of entering a complete URI in the "SIP id" field,
you should only enter a user name.

I believe the "new account" form states something of that sort as an
example right underneath the field.

(Something like that is simply necessary

Agreed.

(I also did not find any possibility to use IPv6.)

That's because IPv6 support in SIP Communicator is completely
transparent. In other words - you don't need to check any options in
order to use it. There is however a problem when directly dialing IPv6
addresses:

https://sip-communicator.dev.java.net/issues/show_bug.cgi?id=577

It might have been the reason for your troubles. I'll be looking into it
tomorrow or on wednesday.

Outgoing calls died in the moment when the other party tried to answer them.

would need to see the logs in order to help

Three other SIP softphones and one hardware phone worked fine under
exactly the same conditions. (Linphone could even establish a video
call using both IPv6 and (tunneled) IPv4.)

No worries, SC can do this too ;). We just need to figure out what's
going wrong there.

Then I enabled a media proxy on my Jabber server and tried to
establish calls via Jabber. (I use the OpenFire server. Don't know
whether its media proxy is compatible or necessary, but switched it
on anyway.) I disabled the Jaber server's firewall completely.
Unfortunately, calls via Jabber behaved the same way as calls via
SIP. Incoming calls died after a few seconds when answered. There was
no audio. Outgoing calls showed Alerting the user and played a
ringtone. From the caller's point of view, they died immdiately when
the other party accepted them.

Let's drop jabber for now. We'll get back to this as soon as we've
completed jingle support.

Well,this was somewhat frustrating...

I completely understand, and appreciate your effort to report to this
list despite your frustration! Hopefully these things will be happening
less often once we get out of the alpha state.

Cheers
Emil

···

Obviously, video remained a
pipe dream when even sound did not work. It seems that Jabber and SIP
work fine, but the RTP stream gets stuck somewhere. :frowning:

It would be great if I could establish direct IP-to-IP calls within
LAN (or IPv6).

The problem would become much easier to diagnose.
Maybe I just misconfigured or misunderstood something.

Regards,

Andrej

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#3

Andrej,

Even i am also facing the same problem, i am being to able hear ringtone
when the other user calls me, but once if i attend the call, thats all
nothing works. no voice. i only tried voice.

i have no firewall between . where is the problem. if in rtp, how to debug
it...

Regards,
HarishKumar.V

···

On Sat, Feb 21, 2009 at 1:01 AM, Andrej Podzimek <andrej@podzimek.org>wrote:

Hello,

I tried to establish some voice (and possibly video) calls using today's
nightly build. Unfortunately, neither voice nor video worked.

I tried both Jabber and SIP.

Using SIP, the client was able to register with my SIP provider and show
incoming calls. When I tried to answer the calls, there was no audio and the
call died after a few seconds. I was unable to find out how to call IP
addresses directly, with no intermediate registrars and proxies. (Something
like that is simply necessary and supported by Linphone or Twinkle, for
example.) (I also did not find any possibility to use IPv6.) Outgoing calls
died in the moment when the other party tried to answer them.

Three other SIP softphones and one hardware phone worked fine under exactly
the same conditions. (Linphone could even establish a video call using both
IPv6 and (tunneled) IPv4.)

Then I enabled a media proxy on my Jabber server and tried to establish
calls via Jabber. (I use the OpenFire server. Don't know whether its media
proxy is compatible or necessary, but switched it on anyway.) I disabled the
Jaber server's firewall completely. Unfortunately, calls via Jabber behaved
the same way as calls via SIP. Incoming calls died after a few seconds when
answered. There was no audio. Outgoing calls showed Alerting the user and
played a ringtone. From the caller's point of view, they died immdiately
when the other party accepted them.

Well,this was somewhat frustrating... Obviously, video remained a pipe
dream when even sound did not work. It seems that Jabber and SIP work fine,
but the RTP stream gets stuck somewhere. :frowning:

It would be great if I could establish direct IP-to-IP calls within LAN (or
IPv6). The problem would become much easier to diagnose. Maybe I just
misconfigured or misunderstood something.

Regards,

Andrej

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