#481 No audio receving at CSipSimple when interacting with a Grandstream GXE-5020 IPPBX

[Anonymous]

Originally created by: mau...@gmail.com

What steps will reproduce the problem?
1. CSipSimple registers with GXE-5020 on the same LAN
2. You make/receive SIP call
3. Audio from CSipSimple reaches the other end OK.
4. No audio to CSipSimple, can't hear the other party

GXE-5020 is interoperating OK with other clients, such as IP phones, X-Pro and SIPDroid. I suspect something is going wrong with incoming RTP at the CSipSimple end.

What version of the product are you using? On what operating system?
0.00-16r410 - Android 2.1-update1 on Motorola Milestone

[Anonymous]

Originally posted by: mau...@gmail.com

More information:

This happens when I call another extension in my same LAN.
If I make call to PSTN audio works both ways, but I hear distorted (ie. robotized) echo at my end. This does not happen with other SIP clients.

[Anonymous]

Originally posted by: mau...@gmail.com

Even more info:

Echo cancellation turned off at CSipSimple end. Problem persists with echo cancellation enabled. If I enable speakerphone audio goes horribly choppy. Finally CSipSimple dies (as reported in previous issue).

[Anonymous]

Originally posted by: r3gis...@gmail.com

Can you try with solution described in issue 449 ?

Maybe linked to psp is also problematic on this device  ( symptoms sounds identical).
If you can also try a reboot of the phone.

[Anonymous]

Originally posted by: mau...@gmail.com

Don't think it's related to issue 449 because from the same LAN I connect to an Asterisk box in the Internet and it works ok. It's funny it works OK doing NAT traversal and fails with this Grandstream PBX at the same LAN.

Issues with Grandstream PBX are divided in two:

1) No incoming audio when calling another SIP client connected to same PBX. Outgoing audio received OK by the other client.
2) Audio in/out when calling through the PSTN trunk, but I hear robotized echo and choppy sound

I would usually suspect from the IPPBX appliance with these symptoms, but the problem is it works ok with other SIP clients other than CSipSimple.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Ok I think you point it out when you said :
"it works OK doing NAT traversal and fails with this Grandstream PBX at the same LAN."

It's cause pjsip sip stack consider one unique "public" IP. As consequence if your SIP server is outside your network or if you activate STUN, pjsip will try to ensure your public IP is the one seen by your sip server. Not the one in your LAN.

If your granstream is configured to not use your SIP server as proxy you'll fall in this problem.

In fact you have several possible configurations with pjsip that may works in all cases.

If your sip server is accessible from LAN. Use DNS to resolve LAN IP of your SIP server when you are in LAN and public IP when you are in public network. As consequence the given "public IP" of your SIP client will always be correct. Also disable STUN (or use it on a STUN server that is on your LAN too). If you can't configure your DNS properly you can also disable the rewrite contact feature of CSipSimple (see FAQ).

If your sip server is on the public area, your should ensure that your(s) access point allow loop back, cause obviously, (unless you call using direct IP), packets will try to go out using external interface (visible by the sip server) and re-enter your network. If your network does not allow that, same problem ...

So in your case I guess only a problem with configuration. NAT / networks problems are really complex, you have to be clear with what you are trying to do (how data packets should go through the network) and how you configure SIP clients.

Getting some logs from csipsimple may helps (see HowToCollectLogs) to know what is the public IP announced by CSipSimple and understand where Grandstream send its media packets (if your ippbx does not media gateway - which would be a good idea to solve all problems).

Mergedinto: 345
Status: Duplicate

[Anonymous]

Originally posted by: r3gis...@gmail.com

Forgot to say, I merge with issue 345, cause if you have 2 sip accounts, you may have this problem cause pjsip only have one public IP. (so even without STUN pjsip will try to resolve its public IP once).