4 Replies Latest reply on Apr 7, 2014 8:41 AM by levi

    RTP Monitoring on Hosted PBX

    fiberman New Member

      We love the Adtan RTP monitoring feature, and it works well especially for voice calls through FXS and PRI ports on the Adtran AOS devices. It gives us detailed call records showing the quality of the call, incoming number, outgoing number, date, time, etc.... I believe it shows all this detailed information since the SIP user account or trunk group is setup on the Adtran device itself, so it grabs all this detail (call to from name etc) from the SIP packet.

       

      We would like to do the same for SIP based voice service that is not terminated directly to the Adtran device, but to phones/devices behind the Adtran unit.


      An example would be Polycom PoE phones plugged in directly to an Adtran Netvanta 1335. We configure the Polycom phones with the external internet sip server, external internet outbound proxy, user id, password, etc. The phones communicate directly through the Adtran and over the internet to the SIP provider. In this scenario we enabled RTP monitoring. For every one call we see two RTP flows (one from the internal phone VLAN and one from the external internet VLAN). The Adtran device rates the call quality, but if you dive down into a call it does NOT show caller ID, number, etc. Is there a proper way to setup the Polycom phones and Adtran where RTP monitoring would be just one per call and show all the detail like it does for a regular voice FXS scenario?


      I assume you would set the phones to the Adtran as the outbound proxy server, and the have the Adtran point those packets to the SIP providers outbound proxy server. The only problem with this is our SIP voice provider auto configures the phones and sets the outbound proxy server to their proxy server. Yes, we could go in an manually change it, but if the phones rebooted it would go back to default. Any recommendations?

       

        • Re: RTP Monitoring on Hosted PBX
          levi Employee

          fiberman:

           

          Thank you for asking this question in the support community.  Here is the Configuring Voice Quality Monitoring in AOS guide for reference.  The VQM will always show two RTP streams, as you mentioned, one in each direction.  Monitoring the RTP streams through the interface provides a method for determining if quality issues, such as packet loss, jitter, and echo, are present in RTP VoIP calls and if these issues are being detected on the private side local area network (LAN), or the public side wide area network (WAN). The statistics gathered reveal if a quality issue is on your side of the connection or on the service provider side of the connection.

           

          Furthermore, if you are using the AOS unit in a SIP proxy application, then if the numbers are involved in the call, you should be able to view them in the VQM output.  However, if the SIP proxy is off, and the SIP ALG is on (by default), then you will not be able to see this detailed information because the AOS unit is simply monitoring UDP packets to see if they are RTP.  Based on the information you provided, you may need to setup SIP proxy transparent.

           

          I hope that makes sense, but let me know if you have any additional information or questions.  I will be happy to help in any way I can.

           

          Levi

          • Re: RTP Monitoring on Hosted PBX
            fiberman New Member

            Levi,

             

            Yes, this makes a lot of sense. I guess I was confused by the two call legs as before we only used VQM for FXS ports or PRI ports, so I guess SIP was only on one side of the call hence one RTP stream right?

             

            I wasn't aware of SIP Proxy Transparent, but it seems like a great option to try. So you are saying if SIP Proxy is not enabled then only Any RTP monitoring option will work, but if proxy is enabled the SIP RTP monitoring should work right?

              • Re: RTP Monitoring on Hosted PBX
                jayh Hall_of_Fame

                fiberman wrote:

                 

                Yes, this makes a lot of sense. I guess I was confused by the two call legs as before we only used VQM for FXS ports or PRI ports, so I guess SIP was only on one side of the call hence one RTP stream right?

                Correct.

                I wasn't aware of SIP Proxy Transparent, but it seems like a great option to try. So you are saying if SIP Proxy is not enabled then only Any RTP monitoring option will work, but if proxy is enabled the SIP RTP monitoring should work right?

                Yes.  Because the Adtran device isn't processing the SIP, it doesn't capture the call setup information with CLID, etc..

                 

                SIP proxy transparent has a number of other features that are worthwhile.  You can track registration status, it cleans up the NAT automatically, and you can do registration pacing.

                 

                Also, any RTP monitoring  can be somewhat of a resource hog.  It looks at all UDP streams so if you have a lot of non-voice UDP traffic it spins its wheels a lot.  With SIP proxy enabled you won't waste CPU analyzing traffic for no reason.

                 

                Note that RTP monitoring only shows the call as seen by the unit.  This will show impairment of traffic reaching the unit towards the phone from the Internet and leaving the phone via the LAN, but if there's an issue outbound on the Internet where the remote party hears bad MOS due to loss/jitter/latency after it leaves the Adtran it has no way to show it.

                1 of 1 people found this helpful
              • Re: RTP Monitoring on Hosted PBX
                levi Employee

                fiberman:

                 

                I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

                Thanks,

                 

                Levi