SIP Trunk- one way audio

[begems]

Hi,

We've a 3300 Cx (version 4.1) using sip-trunks for incoming/outgoing traffic.

In some call scenario's, we have one way audio: on an outgoing call, the called party hears the calling party, but the calling party hears nothing.

After hours of investigation, we found out that the initial RTP port at the phones side changes as soon as the RTP-speech-stream starts. I think the provider is not informed (and he confirms this) about the port change and he's sending the incoming stream to a different port as the outgoing stream.

Anyone had the same problem before? Is there a parameter to let the audio always pass through the controller or to tell the system not to change the port number?

Regards,

Gerry

 

[IrwinMFletcher]

What you describe doesn't make sense. On an outgoing call, if the called party hears the audio, then the internal IP set IS streaming the RTP to the correct port. If the local IP set cannot hear the audio, then the external set is streaming to the wrong place. It doesn't matter what port the internal phone is streaming to.

Can you give more details of your investigation? i.e. what ports are in use and what the SIP messages say.

 

[begems]

Hi Irwin,

It makes sense, in the way that you kind of agreed on the combination phone_ip/portnr towards the SIP provider. That's why you can hear ringback tone. Then, when my speech-RTP starts, the phone sudenly uses another port number without updating the provider, who still sends incoming stream towards the old port.

This is what I saw in the trace, provided to me by the provider (dedicated link). For a single call, I can see 6 RTP streams:

217.X.X.X = providers public address
195.X.X.X = customers public address
10.68.1.1 = dedicated link / provider's site
10.68.1.2 = dedicated link / customer's site
10.165.0.121= phone's ip address

SOURCE DESTINATION

RTP1: 217.X.X.X/7886 195.X.X.X/24638
RTP2: 10.68.1.1/5072 10.165.0.121/50008
RTP3: 10.165.0.121/50008 10.68.1.1/5072
RTP4: 195.X.X.X/24638 217.X.X.X/7886
RTP5: 10.165.0.121/50276 10.68.1.1/5072
RTP6: 195.X.X.X/24638 217.X.X.X/7886

In RTP5, you see the Mitel phone uses port 50276 instead of 50008 before.

Now I changed the sip peer profile parameter: 'Force sending SDP in initial invite - Early answer' to 'Yes'.

Hopefuly we'll use the same combination ip address/portnr for the entine call now for the phone.

Regards,

Gerry

 

[IrwinMFletcher]

I'm not completely following the syntax of what you gave, but consider this: Where a phone stream from, has completely no bearing on where it wants the incoming stream sent to, they don't even have to be the same device (i.e. they can be a completely different IP address, so they certainly don't have to be the same port). What is missing here is the SIP signaling. I would trace the SIP messages coming from the 3300 and see what they say.

Since the far end can hear the audio, then to me that means that they got the second indication (that the local IP set is now streaming from port 50276), but they didn't update from that message where to stream to. Just my guess. Get the sip signaling (use wireshark) and post the results or the capture file.