Login in the alternative AEQ server if working with SIP connectivity some trouble situations are faced

Transcription

1 APPLICATION NOTE Login in the alternative AEQ server if working with SIP connectivity some trouble situations are faced

2 AEQ PHOENIX AUDIOCODECS. APPLICATION NOTE 0-I Login in the alternative AEQ server if working with SIP connectivity some trouble situations are faced There are some scenarios involving several mobile audiocodecs or audiocodecs from different manufacturers, where the SIP protocol is used in order to guarantee the highest compatibility as possible. This protocol solves the required signaling for setting up and realising calls, without worries about the IP configuration of the devices. It also negotiates the coding algorithm used in both ends. This helps to establish a successful communication. In order to fully exploit the SIP advantages, an external SIP server (or SIP Proxy) can be used to establish the calls. The involved users must be registered in this server. AEQ provides two of these servers for free to all their customers. They are accessible through Internet with the following links: sip.aeq.es and asterisk.aeq.es. In order to connect to a server, the customer must have a user name and password. All audiocodecs supplied by AEQ are configured with that user and password (one pair per channel) in order to use with sip.aeq.es. We recommend using this server by default. However, there are some situations where the NAT (Network Address Translation) applied in the local routers doesn t allow a direct communication between the audiocodecs. When this happens, it is usually possible to establish the call between both of them, but audio is not received in one or both devices (not receiving audio status often appears as CONNECTED_NO_DATA ). In these cases, the use of the alternative asterisk.aeq.es server is recommended, as this one always works as an RTP proxy. This means that not only the signaling, but also the audio stream, pass through it. Last but not least, users making an intensive use of SIP communications with many audiocodecs involved are recommended to install their own Asterisk server in order to avoid potential problems caused by bandwidth limitations. 1. PHOENIX AUDIOCODEC CONFIGURATION The following paragraphs describe how to properly configure a Phoenix audiocodec in order to operate with an AEQ Proxy server. Explanations are provided for control Phoenix remote control software, instead of showing how to configure each 1

3 particular model locally (if at all possible), as the software interface is common to all Phoenix devices (except for Phoenix Mobile). Configuration for sip.aeq.es server is already made in factory in all AEQ audiocodecs. You only need to choose the SIP, Proxy based option in the INTERFACE drop-down menu. Once the main control window is accessible for the device, click on the CONFIG button, at its right side, and select the GENERAL option. A configuration window similar to the one displayed in figure 1 will shows up: Figure 1 Once in this general configuration window, click on the I/F Setup option for the channel we are trying to configure: 2

4 Click on the SIP Parameters within the window that appears then: 3

5 Next, select Manage Provider to add a new server: Select New Provider: 4

6 You must fill the Host and Domain fields in with asterisk.aeq.es and, then, a symbolic name for this server (i.e: Asterisk ) must be provided in the Description field: Once the new provider is created, close the window. Now, it should appear in the list of SIP providers: 5

7 Next step: make sure to select this new provider and add the user names, password and domain. These details will be provided by AEQ under request (the application may fill the same value for the User name and User fields meanwhile you type the first one in). The Display Name will only be used to identify the incoming calls: Click on the Apply button in this window and in the former one (I/F Setup): 6

8 Now, the audiocodec should be ready to use Asterisk as its SIP server. Please double-check that the channel status changes to OK (if it shows REGISTER_ERROR or REGISTERING as in the following figure during more than a few seconds, you will need to check the SIP configuration specially that the user and password are typed correctly- as well as the device IP configuration to make sure that it has Internet access -check that the IP address, mask, gateway and DNS server are properly configured-. Once two audiocodecs are successfully registered in Asterisk, you can check whether it is possible to make calls from one to another and that audio is received in both ends. 2. REMARKS Please note that the users and passwords provided by AEQ for the alternative Asterisk server have nothing to do with the ones provided (and pre-configured from factory) for registering in sip.aeq.es. This is because Asterisk only admits numeric users. Not all coding algorithms implemented by the audiocodecs can be used with the Asterisk server. In particular, it doesn t support AEQ-LD, MPEG-2 or AAC-LC/LD. However, the ones that are should, nevertheless, cover most users needs: G.711, G.722 and several OPUS MUSIC modes with different binary rates in stereo (64, 128 and 192kbps), and mono (at 64 kbps). 7