[One way audio]

Is there a public version of pbxnsip that changes the SSRC when the call is answered? I have only see wiresharks traces of it. I have made some changes and it would be verify that the issue is fixed.

[One way audio]

I have another question

 

How come the SSRC changes, when I answers a call? (please note that all codec's and sdp session Id is the same)

 

I cannot find anywhere where this isent allowed, but what is the point in creating a new RTP instance?

 

The only thing I can find in rfc-3550 is:

"Because the randomly allocated SSRC identifier may change if a

conflict is discovered or if a program is restarted"

 

I am 100% sure that the SSRC's are different and that the phones hasent restarted.

[One way audio]

There is a checklist at http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems that might be a good point to start with.

 

Hi,

 

Thx that worked. I just had to disable STUN

 

Best Regards

Rune

[One way audio]

Hi,

 

I have a problem with one way audio when I recieve incomming call on the trunk.

 

My setup:

The pbxnsip is placed on a LAN behind a DLINK 603 router. On the same network I have 3 sip-phones.

The pbxnsip has a registration on musimi.dk. The trunk is configured to forward incomming call to extention "2222".

The 3 sip phones has each a registration on the pbxnsip (phone number 1111,2222,3333).

 

If I make an incomming call to the musimi account (the trunk on pbxnsip), phone 2222 alerts, when I answer there is no TX audio from phone 2222 to the external device. Any hints why? Does the pbxnsip requires a public IP address to route the RTP correctly? What NAT traversal features does pbxnsip have? I cannot find a setting for rport or ice.

 

I have made an wireshark trace and audio from the extention unit is routed correctly to the pbxnsip (I have checked ports, ip-addresses).

 

If I move the trunk SIP account to another phone on the LAN and make an incomming call there is audio both ways (I have enabled rport on the device). Likewise if phone 1111 calls 2222 (internal call on the pbx) audio is also working fine.

 

Another issue is that if the trunk registration has multiple bindings on the external SIP server (in this case musimi) the pbx cancelles the incomming call when the all is answered.