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No RTP coming out of SnomONE server


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Im having a global issue where a number is dialed, the extensions phone counter starts, but there is no audio. When the call is disconnected, it disconnects.

 

SIP is going through fine, but not RTP. This is reflected in the PCAP.

 

During the call, there is no audio on either side, but it does ring the called party. The pcap has only the outside audio stream. In other words when I listen to the call, there is no audio coming from the internal extension, just the ringing and the called party answering saying, "hello...hello...CLICK"

 

As I stated, the pcap shows SIP going back and forward, but only inbound RTP. There is NO RTP coming out from the server.

 

The pcap was taken from the SnomONE server.

 

The inbound audio is hitting the right RTP port.

 

Im running Version: 5.0.10 (CentOS64)

 

Anyone have any ideas? Im at a loss and ive got angry customers.

 

 

 

 

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it may be hitting the RTP port but could be sending the packets to a different IP.

​please cross reference any of the scenarios found in this article http://wiki.snomone.com/index.php?title=One_Way_Audio

For example is the PBX sending RTP to VOIP provider and is the VOIP provider sending the RTP traffic to the PBX.

 

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Thanks for that.

 

I should still be seeing the outbound RTP stream, correct? The pcap was from the server and I would expect to see the outbound RTP at least attempting to go out. But I do not see any outbound RTP from the server, just inbound from the outside party. This is confirmed in the pcap as well as I only hear the outside parties audio stream, and see only inbound RTP.

 

As far as the inbound RTP stream goes, it looks good. It hitting a good rtp port (57806) using ULAW. The server has only 1 IP address. The outbound proxy is configured on the trunk. I dont think its a firewall issue as it is not persistent. They call and it fails. They call again and it works.

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Not sure if this is important, but something Ive noticed.

 

All of these phones are remote. This being the case, all phones on a domain use the same (external) IP to register. They are differentiated by port number.

 

Looks like some if not all of these phones are using ports in the RTP port range.

 

Im not sure if this info is pertinent. Hate to muddy the water...

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We are not living in an ideal router world. Many routers have limitations on how many NAT connections they can have at the same time. You might just run out of them. Usually the tables are separate for TCP and UDP, so that you might have the situation that you still have enough TCP connections thus the registrations are stable, but you are running out of UDP, thus you have problem with media.

 

The other problems is when routers are firewalls and filter stuff out that they don't understand. Check if there is anything with "SIP" or "ALG" on the router and turn it off.

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I will check the router and get back to you.

 

Question - Even if the router was out of UDP sessions, wouldn't I still see the RTP stream attempting to go out. Wouldn't it be on the router that the stream gets dropped?

 

Im wondering why im not seeing the stream when the pcap was taken right off the PBX?

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I would say 90 % this is a routing problem. The PBX is probably supposed to send the media to an address that it cannot reach, for example a hosted PBX sending to 192.168.1.100. You will also see this easily in the PCAP or the trace.

Can I send you the pcap in a PM? Im a little confused as there is no outbound rtp to see what IP its trying to send to.

 

Looking at the pcap, I only see the pbx ip, and my upstream provider. I would expect the rtp to be coming from the pbx ip, but alas, there is no outbound audio at all.

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  • 3 weeks later...

Just to give everybody a heads up. The issue seems to be related to multiple 18x responses coming back with the same To-tag in them and the same SDP, but slightly different message content so that they are not treated as message repetition. Workaround can be to set the trunk to ignore the SDP in the 18x messages. Next version will contain a fix to suppress repetitions completely.

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Just to give everybody a heads up. The issue seems to be related to multiple 18x responses coming back with the same To-tag in them and the same SDP, but slightly different message content so that they are not treated as message repetition. Workaround can be to set the trunk to ignore the SDP in the 18x messages. Next version will contain a fix to suppress repetitions completely.

How do you set the trunk up like that?

I assume you mean on the PBX settings.

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There is a setting called Ignore18xSDP, it is only visible in the text editing mode. Set it to true, make sure that you enter either the right password in the text or take the password ****** field out and then hit the save button.

Will this require a restart of the service?

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There is a setting called Ignore18xSDP, it is only visible in the text editing mode. Set it to true, make sure that you enter either the right password in the text or take the password ****** field out and then hit the save button.

And what password? I assume the trunk secret but want to be sure.

 

Ive got a lot of customers on this box so I want to make sure Im understanding.

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