I can't hear audio, whats wrong?

Article Details

URL:	http://support.gizmo5.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=354
Article ID:	354
Created On:	02 Jan 2008 01:24 PM

Answer

The most common cause of a bi-directional audio issue (no sound or one-way sound) is due to a misconfigured Firewall/NAT/Router (gateway device). This article will assist you in configuring your gateway device for use with different variety of clients or multiple clients that can be used with Gizmo Project.

The symptom ususally experienced of bi-directional audio is usually caused by a gateway device that has network ports blocked or another adverse configuration that prevents the signal from returning to your client. A few examples are firewalls that modify or inspect packets (External NAT or Stateful Packet Inspection). In most cases, the Gizmo Project SIP network will detect your network setup and adjust automatically, but in some cases the network setup will not work without end user configuration. Most issues relating to login issues, and calls dropping can be caused by this Firwall/NAT related issue as well.

Yes there are some other soft clients that may work behind highly restrictive networks, but in some ways they are not using SIP or a highly modified proprietary version of SIP. They tend to use port-hopping methods probing for an open port and finally tunneling though port 80 http as a last resort. These can be a vulnerability to that user and that network if compromised. We choose to use the open standard SIP protocol for an "open" network.

Possible solutions to this scenario as well as an explanation are as follows:

For SIP communication to occur in your network, two types of traffic streams must be able to enter your network and then be routed to your device. They are SIP for Signaling the client, RTP for the actual media and RTCP for the media control and negotiation.

Gizmo Project Client:
The Gizmo Project client uses the specific ports of 64064 for SIP, 5004 for RTP and 5005 for RTCP. These ports will need to be opened on your gateway device if possible or port forwarded to the IP address of your client.

Hardware Devices:
Other Clients such as an ATA or WiFi Mobile device may have other requirements for SIP communication. It is common for most client devices to use the standard SIP port 5060. For RTP and RTCP however, it is possible that by default your device will choose a random set of ports to listen to for the return media. On some devices this can be configured to listen to specific ports such as 5004-5005, but if you do not see this option, it will be necessary to open a broader range of ports in order to accept the possible random ports. We commonly suggest ports 10000-20000 but it can vary depending on your hardware provider. Please check with your hardware manufacturer to determine the default ports that you device listens for RTP and RTPC (usually next odd port from RTP).

Multiple Clients:
Each additional client behind the same network would require the next consecutive ports to be opened for SIP RTP and RTCP.

Troubleshooting:
We also highly suggest that if possible, connect your client device or computer directly to the Internet Modem, essentially bypassing your Firewall/NAT device to isolate a possible NAT traversal Issue. It is also important to verify that you have the most up to date software or firmware for your client to ensure compatibility with current SIP standards.

NAT Traversal Solution:
Other NAT issues can be resolved by using the STUN protocol. The desktop client has it configured by default but if your device has an option for it, enable it and use stun01.sipphone.com as the server and the default port of 3478.

As a last resort, it may be possible that the ISP is not allowing SIP/VoIP traffic so it may be a good idea to contact them and verify as well.