[Spce-user] Hair-pinning through SPCE - Loss of RTP Traffic

[Gary Nieboer]

We are having an issue with calls that pass through the SPCE and hairpin
through a FreePBX back through the SPCE and out again.  (Our FreePBX is set
up to Call Forward Always to a PSTN number)   SPCE is connected to Carrier
A  peer, and the outbound default peer.  SPCE is connected to FreePBX as a
peer with a few peering rules.

This only seems to happen when all 4 call legs to peers are passing through
the same SPCE and 2 of the legs are to the same FreePBX.

SIP signaling works great.
We have no RTP in either direction on all 4 legs.

Call Leg A:   Carrier A peer to SPCE
Call Leg B:    SPCE to FreePBX-A peer
Call Leg C:    FreePBX-A peer  to SPCE
Call Leg D:   SPCE to Carrier A


Other items to note:


On all peers:  "use_rtpproxy"  is set to "Always with Plain SDP"

Call Leg D can also connect to a different Carrier, and we get the same
missing RTP.

Carrier A & alternate Carrier tested do not Proxy Media.

Legs A & B alone work fine

Legs C & D alone work fine

The combination of all 4 legs, causes a loss of RTP in all legs.  (Note:
 If no audio comes from the carriers on Legs A & D, we would expect no
audio/RTP on legs B & C)

As an alternative test, we also adjusted CID in the FreePBX so we sent out
a different "caller" on legs C and D.  This also resulted in lack of RTP
media.


SDP messages appear to indicate correct communication of IP addresses and
ports for media flow, and we do not show any indication of firewalls
blocking the applicable udp traffic.



I'm not sure where to look next.  Any advice/ideas will be greatly
appreciated.




Additional testing information:


In all of the below, the FreePBX Peer's Inbound Route (DID) is set up to
CF-A to PSTN Phone B telephone number.


*In the below two scenarios, we have No RTP traffic in either direction.
SIP signaling works fine and call is dropped after 30 seconds without RTP*


PSTN Phone A  >  Carrier A Peer   >  SPCE-A >  FreePBX Peer > SPCE-A >
 Carrier A Peer  >  PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >  SPCE-A >  FreePBX Peer > SPCE-A >
 Carrier B Peer  >  PSTN Phone B



*SIP and RTP work great in all of the below instances:*

PSTN Phone A  >  Carrier A Peer   >   FreePBX Peer > SPCE-A >  Carrier A
Peer  >  PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >   FreePBX Peer > SPCE-A >  Carrier B
Peer  >  PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >  SPCE-A >  FreePBX Peer >  Carrier B
Peer  >  PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >  SPCE-A >  FreePBX Peer >  Carrier A
Peer  >  PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >  FreePBX Peer >  Carrier A Peer  >
 PSTN Phone B

PSTN Phone A  >  Carrier A Peer   >  FreePBX Peer >  Carrier B Peer  >
 PSTN Phone B




In the below instances, the FreePBX Peer is set up to ring an extension
when the DID is called.

*SIP and RTP work great in all of the below instances:*

PSTN Phone A >  Carrier A Peer > SPCE-A  >  FreePBX Peer > SIP Phone
(Extension)

SIP Phone (Extension) > FreePBX Peer  >  SPCE-A > Carrier A Peer

SIP Phone (Extension) > FreePBX Peer  >  SPCE-A > Carrier B Peer



Thanks,

Gary Nieboer
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