Re: [asterisk-users] RTP over TCP
SIP is just the control protocol, and can be negotiated over TCP or UDP. The
actual payload is done over RTP, which is a UDP-based protocol.
If you had to add firewall exceptions/PAT config for the TCP SIP traffic,
you'll also need to add the same for RTP traffic as well.
-- Nathan Clemons
On Fri, Apr 23, 2010 at 12:21 PM, wrote:
> Hi List,
>
> i have to put an * between two other SIP gateways and due to some
> circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
> fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
> (ocs) and that's about it. In the other direction however (ocs -> me ->
> deverto4) the call setup is complete but there is no audio.
>
> I can see the audio in the form of tcpdump, but neither party hears the
> other side. Tcpdump also shows that while the call setup is via tcp,
> the audio is transmitted via udp. I'm guessing this is the reason of
> silence. The first setup is working because one of the gateways are
> supporting sip over tcp only and * accepts both.
>
> my setup is pretty simple as * is only handing calls over to the
> gateways. Relevant parts are below.
>
> could anyone please confirm that it is an error, that asterisk sends the
> RTP stream via udp and this is the cause of the silence? Is there any
> way to tell asterisk to use tcp only? I'm aware of the drawbacks, but i
> still need to get this working.
>
> I'd appreciate any help.
>
> thanks
> adam
>
>
> sip.conf:
>
> tcpenable=yes
> tcpbindaddr=0.0.0.0
>
> [ocs]
> type=friend
> host=192.168.1.1
> context=ocs
> qualify=yes
> transport=tcp
> nat=no
> canreinvite=no
> disallow=all
> allow=alaw
> allow=ulaw
>
> [deverto4]
> type=friend
> host=172.18.200.4
> context=deverto
> qualify=yes
> nat=no
> canreinvite=yes
> transport=tcp
> disallow=all
> allow=alaw
> allow=ulaw
>
> and the extensions.conf:
>
> [deverto]
> exten => _X.,1,Dial(SIP/${ext...@ocs)
> exten => _+X.,1,Dial(SIP/${ext...@ocs)
>
> [ocs]
> exten => _X.,1,Dial(SIP/${ext...@deverto4)
> exten => _+X.,1,Dial(SIP/${ext...@deverto4)
>
--
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Re: [asterisk-users] Testing a sip call through Asterisk?
Turning on qualify=yes, or qualify=60, seems to break the BroadVoice connection (it goes from UNKNOWN to UNREACHABLE and calls fail). I'm wondering if they don't support OPTIONS probing or something. -- Nathan Clemons On Fri, Apr 16, 2010 at 3:22 PM, Jeff LaCoursiere wrote: > > > On Fri, 16 Apr 2010, Nathan Clemons wrote: > > > I'm looking to find a test tool that will register with our Asterisk > > (Trixbox) server here at work and place an outgoing call via our main SIP > > trunk (BroadVoice) to confirm that things are working. I've looked around > > but I can't seem to find any tools that will do what I'm looking for. > > > > I can't just monitor the status of the trunk inside Asterisk, as this is > the > > normal status: > > > > [snip] > > just add "qualify=yes" to your context and it will monitor the RT latency. > > j > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Testing a sip call through Asterisk?
I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: asterisk*CLI> sip show peers Name/username HostDyn Nat ACL Port Status BroadVoice/425256 147.135.32.221 N 5060 Unmonitored ... 37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0 offline] asterisk*CLI> Alternatively, any suggestions as to how I can change the trunk configuration so that it is monitored would be appreciated. The peer config is set as: allow=ulaw disallow=all canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=425256 host=sip.broadvoice.com insecure=very nat=yes secret=XX type=peer username=425256 Any assistance would be appreciated. I'd rather know when things fail via an automated system rather than learning it's down from the users. -- Nathan Clemons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
