So, I've decided to give WiFi calling a spin (despite having a solid 4 bars of sweet, sweet HSPA+ sitting at my desk). I'd like to share my findings with you guys.
Firstly, I've looked into the bandwidth used by WiFi calling, by watching my router's information page.
Here is a screenshot of an active call
The very first section is me tapping on the mic, while waiting on hold with HTC (that's a whole 'nother story). The second part, silently listening to their boring hold music/we appreciate your call as a valued customer crap, the third part, a two-way, engaged conversation, and finally, me giving her my name, ticket number, etc, while she silently listened.
By the way, despite the green line being stated as "in", it is actually the outbound (upstream) bandwidth from my WAN interface. It's reporting as "in" because the data is flowing out of the WLAN chip on my phone, in to the WLAN interface (wl0) on the router, then going through the bridge (br0), and out the WAN port (vlan2) on my router. And, in effect, the red line would be incoming (downstream) data on my WAN, and it is flowing out of wl0 into my phone. Hope that makes sense. Also, my computer that I was viewing this info from did not fudge the results at all, I am connected via wl1 (the 5 GHz interface), and my G2 was the only client on the 2.4 GHz interface.
During this call, there was also some other traffic flowing in/out of the WAN interface (traffic from my PC, broadcast messages, and what not). Nothing extreme. The maximum bandwidth used at any point in the call was about 80kbps (kilobits per socond) in either direction. I am going to go out on a limb here and guess latency is more important to call quality than available bandwidth.
I do not have packet-level QoS enabled in my router, however, I do have WMM/WME (Wireless Multimedia Extensions) enabled, with DD-WRT's default settings. I am unsure if Kineto's UMA app tags the frames with the necessary information for it to be prioritized by WMM/WME. I guess I will look into that later.
From a subjective point of view, the woman I was speaking with sounded crystal clear, quite possibly better than a typical UMTS/GSM based call. I also asked her how I sounded on her end, and her response was "absolutely gorgeous". Then, we got back to the actual conversation, and about 30 seconds later, the call dropped. Hmph. The call drop log in SETTINGS --> ADVANCED SETTINGS --> CALL DROP LOG is as follows:
Perhaps we can find reference to all of the error codes somewhere, and find out exactly what went wrong?
Well, that just about sums up my findings and information for now. I will be looking deeper into this, and how we can make it better, and more stable. I will also try to keep this post updated with my findings, as well as those of others, so you do not have to read through many pages to find all of the information.
Firstly, I've looked into the bandwidth used by WiFi calling, by watching my router's information page.
Here is a screenshot of an active call
The very first section is me tapping on the mic, while waiting on hold with HTC (that's a whole 'nother story). The second part, silently listening to their boring hold music/we appreciate your call as a valued customer crap, the third part, a two-way, engaged conversation, and finally, me giving her my name, ticket number, etc, while she silently listened.
By the way, despite the green line being stated as "in", it is actually the outbound (upstream) bandwidth from my WAN interface. It's reporting as "in" because the data is flowing out of the WLAN chip on my phone, in to the WLAN interface (wl0) on the router, then going through the bridge (br0), and out the WAN port (vlan2) on my router. And, in effect, the red line would be incoming (downstream) data on my WAN, and it is flowing out of wl0 into my phone. Hope that makes sense. Also, my computer that I was viewing this info from did not fudge the results at all, I am connected via wl1 (the 5 GHz interface), and my G2 was the only client on the 2.4 GHz interface.
During this call, there was also some other traffic flowing in/out of the WAN interface (traffic from my PC, broadcast messages, and what not). Nothing extreme. The maximum bandwidth used at any point in the call was about 80kbps (kilobits per socond) in either direction. I am going to go out on a limb here and guess latency is more important to call quality than available bandwidth.
I do not have packet-level QoS enabled in my router, however, I do have WMM/WME (Wireless Multimedia Extensions) enabled, with DD-WRT's default settings. I am unsure if Kineto's UMA app tags the frames with the necessary information for it to be prioritized by WMM/WME. I guess I will look into that later.
From a subjective point of view, the woman I was speaking with sounded crystal clear, quite possibly better than a typical UMTS/GSM based call. I also asked her how I sounded on her end, and her response was "absolutely gorgeous". Then, we got back to the actual conversation, and about 30 seconds later, the call dropped. Hmph. The call drop log in SETTINGS --> ADVANCED SETTINGS --> CALL DROP LOG is as follows:
Code:
CD-12 ISP Problems
2010:11:04:11:06:18:336Well, that just about sums up my findings and information for now. I will be looking deeper into this, and how we can make it better, and more stable. I will also try to keep this post updated with my findings, as well as those of others, so you do not have to read through many pages to find all of the information.
