one way audio for incoming calls

[jokiin]

I was going to change the extension and attach the config file. But it seems with win 7, access to the file extension has been removed. I often find I have the question why?

control panel/folder options/view

[Baldrick]

Fosfet,
I see you have tried defaulting your pap2 unit. I wrote a program a couple years ago to configure these Linksys units very quickly. Basically you input your lan ip, number, pass, state, etc & it sets up all your sip, ntp, dialling rules, tones, codec options, keep alive, etc for you & I would be game to bet negate the need for any port forwarding in your router.
Would you like for me to put it on my personal webspace so you can download & try. ( Maybe it will not help your situation, but may just be worth a try...)

[fosfet]

The upload was rejected because the uploaded file was identified as a possible attack vector.

So changing the extension did not work.


Hi Baldrick,

reset the router, not the ata. Lodperera wisely noted that as I am also having issues with xlite, my router is indicated. At the moment. To re-cap, it did work for a few years.

thanks, tim

[jokiin]

The upload was rejected because the uploaded file was identified as a possible attack vector.

So changing the extension did not work.

just zip the file, no need to rename it

[fosfet]

pap_config.zip

(15.52 KiB) Downloaded 127 times

thanks, tim

[lodperera]

Hi fosfet,

Just checked your config.


Use preferred as G711a and disable all the other codecs for testing.

Try changing in SIP--> NAT support Parameters.

Handle VIA received: yes Handle VIA rport: yes
Insert VIA received: yes Insert VIA rport: yes

-----------------

Even that fails try to take a call and copy the

RTP Packets Sent: RTP Bytes Sent:
RTP Packets Recv: RTP Bytes Recv:

so we could check whats really going on.

[Dazzled]

Fosfet, please don't be distracted from what lodperera needs for analysis, but you might be interested in VoIP monitoring with Wireshark (http://www.wireshark.org/). A Windows example intercepting a softphone is shown at http://www.unappel.ch/public/100119-wireshark-xlite/. There are many more examples and videos out there, as well as the Wireshark documentation. Intercepting the ATA needs suitable switch or hub hardware in your network.

[fosfet]

Try changing in SIP--> NAT support Parameters.

Handle VIA received: yes Handle VIA rport: yes
Insert VIA received: yes Insert VIA rport: yes

OK, did that.

Use preferred as G711a and disable all the other codecs for testing.

Ran 4 tests

1 711a only, as above
2 All codecs enabled
3 711a only
4 All codecs enabled


First 2 tests, one way audio, and this is reflected in the rtp stats
Second 2 tests, everything ok, and this is reflected in the rtp stats

Thanks dazzled, I'll do some reading.

tim

G711A only enabled 1

Current Time: 1/2/2003 01:36:03 Elapsed Time: 00:33:52
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 436 Broadcast Bytes Recv: 41505
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 3009 RTP Bytes Sent: 421520
RTP Packets Recv: 426 RTP Bytes Recv: 8520
SIP Messages Sent: 173 SIP Bytes Sent: 68734
SIP Messages Recv: 302 SIP Bytes Recv: 63413
************************************************************************************

All codecs enabled 2

Current Time: 1/2/2003 01:47:31 Elapsed Time: 00:45:20
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 581 Broadcast Bytes Recv: 55103
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 4738 RTP Bytes Sent: 698160
RTP Packets Recv: 426 RTP Bytes Recv: 8520
SIP Messages Sent: 241 SIP Bytes Sent: 95640
SIP Messages Recv: 411 SIP Bytes Recv: 89529

Mapped SIP Port: 5060
Call 1 State: Connected
Call 1 Tone: None
Call 1 Encoder: G711a
Call 1 Decoder: G729a
Call 1 FAX: No
Call 1 Type: Inbound
Call 1 Remote Hold: No
Call 1 Callback: No
Call 1 Peer Name:
Call 1 Peer Phone: 0410711140
Call 1 Duration: 00:00:18
Call 1 Packets Sent: 916
Call 1 Packets Recv: 0
Call 1 Bytes Sent: 146560
Call 1 Bytes Recv: 0
Call 1 Decode Latency: 0 ms
Call 1 Jitter: 0 ms
Call 1 Round Trip Delay: 0 ms
Call 1 Packets Lost: 0
Call 1 Packet Error: 0
Call 1 Mapped RTP Port: 16400 >> 0

************************************************************************************
G711A only enabled 3

Current Time: 1/2/2003 01:55:00 Elapsed Time: 00:52:49
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 684 Broadcast Bytes Recv: 64933
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 6077 RTP Bytes Sent: 912400
RTP Packets Recv: 1350 RTP Bytes Recv: 156360
SIP Messages Sent: 283 SIP Bytes Sent: 112673
SIP Messages Recv: 481 SIP Bytes Recv: 105297

Mapped SIP Port: 5060
Call 1 State: Connected
Call 1 Tone: None
Call 1 Encoder: G711a
Call 1 Decoder: G711a
Call 1 FAX: No
Call 1 Type: Inbound
Call 1 Remote Hold: No
Call 1 Callback: No
Call 1 Peer Name:
Call 1 Peer Phone: 0410711140
Call 1 Duration: 00:00:18
Call 1 Packets Sent: 929
Call 1 Packets Recv: 929
Call 1 Bytes Sent: 148800
Call 1 Bytes Recv: 148640
Call 1 Decode Latency: 60 ms
Call 1 Jitter: 2 ms
Call 1 Round Trip Delay: 0 ms
Call 1 Packets Lost: 0
Call 1 Packet Error: 0
Call 1 Mapped RTP Port: 16402 >> 0
************************************************************************************
All codecs enabled 4

Current Time: 1/2/2003 02:22:57 Elapsed Time: 00:00:37
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 12 Broadcast Bytes Recv: 1104
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 950 RTP Bytes Sent: 152000
RTP Packets Recv: 948 RTP Bytes Recv: 151680
SIP Messages Sent: 8 SIP Bytes Sent: 4336
SIP Messages Recv: 9 SIP Bytes Recv: 4402

Mapped SIP Port: 5060
Call 1 State: Connected
Call 1 Tone: None
Call 1 Encoder: G711a
Call 1 Decoder: G711a
Call 1 FAX: No
Call 1 Type: Inbound
Call 1 Remote Hold: No
Call 1 Callback: No
Call 1 Peer Name:
Call 1 Peer Phone: 0410711140
Call 1 Duration: 00:00:19
Call 1 Packets Sent: 952
Call 1 Packets Recv: 950
Call 1 Bytes Sent: 152320
Call 1 Bytes Recv: 152000
Call 1 Decode Latency: 60 ms
Call 1 Jitter: 2 ms
Call 1 Round Trip Delay: 0 ms
Call 1 Packets Lost: 0
Call 1 Packet Error: 0
Call 1 Mapped RTP Port: 16438 >> 0

************************************************************************************

[lodperera]

Hi fosfet,

Seems that the changes were made are taking effect and the issue was resolved by using the G711a which your mobile service provider is forcibly using.

Thank you for sharing detailed information with the community.

[fosfet]

Well thanks Lodperera,

As the system always did work sometimes, I am not quite on board with you at this stage. But I will let you know how it works out. Hope you are in for the long haul as I am going to be out of town for a few months soon. So it will be a while yet before I can share your optimism

tim

[fosfet]

Wed morning, no audio in.

I think my next steps would be to set a fixed IP for the ata, though this seems a little desperate as the router always allocates the same address anyway. And get xlite working. I see with a call coming in xlite returns an address ending in 5060. So I suspect xlite uses port 5060 and therefore I would set up a port forward for that.

However I have run out of time to fiddle with this now. Had hoped to have it sorted before I went away, but now I will have to get back to it when I return in September.

Was also intending to cancel the service for that period, so I will not be able to post on this forum during this time.

thanks for the inputs, tim

[lodperera]

I think my next steps would be to set a fixed IP for the ata, though this seems a little desperate as the router always allocates the same address anyway. And get xlite working. I see with a call coming in xlite returns an address ending in 5060. So I suspect xlite uses port 5060 and therefore I would set up a port forward for that.

Well X-lite uses 5060 for signalling. Which means that if you can receive a call you dont need to port forward that.
But as you are having a one way audio where you cant hear the caller which probably means that you should port forward the RTP 10,000 to 20,000 which carries the media.

PS - If you cancel the service most probably you will not be able to get the same number.

[fosfet]

Hi All,

An update.

I am back, got a new voip account, had so much trouble I came very close to cancelling it again, but now things are looking up........

I guess to cut a long story short, I put a few firewall rules in as recommended by the manual for voip, modified to suit. And that is so far working.

I do not know why the system ran well for years, and now I have to modify the firewall, but assume something external to my gear must have changed.

The (firewall) mods included a provision for port 5060, though as the phone always rang in or out I do not think that this was ever the issue,
As I understand it, the ATA initiates a TCP data exchange between itself and the sip server, so the firewall will remember this and allow free exchange between the two on port 5060

Restricting the RTP ports to a choice of 5 in the ata, and then "allowing" any udp packet access to these five ports destined for the ata,
and "allowing" port access to the ata where the request originated from the ata.
As I understand it, the RTP packets are UDP, and what is actually carrying the voice, and are exchanged between the ATA and some other server (meaning not the sip server). And herein lies the problem, as the firewall has no record of this other server.

Please feel free to correct anything.

Assuming everyone has forgotten, this is for an Ericcson W35 hspd router, and a Linksys ATA, predating the Pap2 T. Maybe a PAP 2 NA


Other stuff along the way;

While i was away, xlite stopped working and insisted that I upgrade. I think it stopped working because faktortel blocked access to external (from aust.) ip's But I did not know this and upgraded xlite. I have never been able to get it working since, and it keeps demanding that I join some Canadian idea. I did point out to both faktortel and exetel that their setup guides are hopelessly outdated for xlite, but nobody took the hint and fixed their setup guides. A lack of action I can completely understand. And given the "attitude" of v 5 xlite, I really do not want to use it in any case.

The ATA call quality was getting a lot of complaints from the other parties (outgoing voice) I ended up reinstalling the factory defaults and took a minimalist approach to getting it going again. This fixed it.


tim

[NIrmitha]

Hi fosfet,

I will escalate your issue to our VOIP engineers. They will get back to you as soon as possible.

[Dazzled]

fosfet, the explanation above is not quite right - there's a good description of SIP at http://www.siptutorial.net/SIP/command.html onward. The whole point of the rigmarole is to get a direct computer/ATA to computer/ATA audio conversation going (in RTP) without being caught up by the basic router firewall rule of no inbound connections unless previously established by an outbound request. By the time you get to the RTP step that state has been established and the router should be happy.

It could be interesting to see the low level firewall rules of the Ericsson. I noticed that the manual refers to the telnet interface, but does not give the default root password. Do you know what it is? If you can get in by telnet, the shell command iptables -L will print out the firewall rules. Each is applied in turn, until you eventually get to a default rule to drop everything. I'd expect to see up at the top provision to accept UDP from anywhere if it is on destination port 5060, and forward it on to the ATA address. There is no real need to have a specific rule for any other port, but it's a good way to stir up trouble.

Regarding softphones, you could try two others that are free - Linphone (http://www.linphone.org/) or Ekiga (http://ekiga.org/) These run on both Linux and Windows. I've found both perfectly satisfactory when installed on laptops. An Exetel customer posted the simple Ekiga config setup eons ago at https://sites.google.com/site/davidtang ... exetelvoip. Free phones don't have G.729 so you will have to use G.711a when you configure. It is also called PCMA in the config pages. With Ekiga, if there is already an account for ekiga.net, delete it.