Voice Quality Issues and RTP Header Compression

Has anyone had problems with RTP Header Compression. A customer is running Voice over IP over our MPLS VPN. They are experiencing packet loss and the interfaces are incrementing flushes and output drops. I have run SAA and seen some SAA packets dropped also. The circuits tested clean so I am beginning to suspected RTP Header Compression. It is enabled on all interfaces and it causes the router to process switch so I am thinking this is creating delays and leads to the flushes and the drops.
Am I on the right path?
Has anyone else seen a problem with RTP Header Compression?
Bruce

hi
do confirm whether you have got any qos policies defined for different traffic patterns or not ?
and also about the traffic patterns being pumped on that link like the utilisation and the applications being passed on..
regds

Similar Messages

7921-CME Voice Quality Issue

Hello,
I am running into an issue at customer site which is going on for over 30 days.
They have consistent voice quality issue with 7921 phones.Have checked their phone webpage and see that jitter value increases to 500 during issue. Sometimes I see tcpretransmissions too,
They have QoS but that doesnt seem to help.
I have even collected sniffer off wireless but not able to play due to wireless header.
Have worked with TAC but they do not find any thing conclusive.Can anyone pls tell me what else can be done to analyse and fix the issue

Is the issue with unicast RTP or with multicast (i.e. paging)?
What AP platform and version are you using?
1.2(1) for the 7921 was just released.
1.2(1) Software
http://tools.cisco.com/support/downloads/go/Model.x?mdfid=280808676&mdfLevel=Model&treeName=Voice%20and%20Unified%20Communications&modelName=Cisco%20Unified%20Wireless%20IP%20Phone%207921G&treeMdfId=278875240
1.2(1) Release Notes
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7921g/firmware/1_2_1/english/release/notes/7921_12.html
Other than that ensure the deployment is inline with the 7921G Deployment Guide.
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7921g/6_0/english/deployment/guide/7921dply.pdf

Quality Issues and Green Strip at Right of Frame in Comp. 3.5

I just upgraded to Compressor 3.5 with my upgrade of FCS which includes FCP 7 (I guess this is FCS 3 even though the upgrade box doesn't say so). The quality of Compressor's outputs using the same settings is noticeably inferior to the previous versions of Compressor in the FCS releases that included FCP 5 and FCP 6. I also have this green stripe that appears at the extreme right edge of the frame.
I mostly do h.264 compressions for streaming. In previous versions of Compressor included with FCS 5 and FCS 6, Compressor made incredibly good quality compressions when set to a restriction of 300 kbps for a 500 x 282, 16.9 screen size using the 'best quality' multi-pass encoding setting. In version 3.5 of Compressor, the quality, using the same settings, is absolute crud and I have this green stripe appearing on the extreme right edge of the frame.
My original source content is usually HDV 1080i60. The problem with my Compressor outputs w/ the green stripe is consistent, however, with footage I've been working with in Apple ProRes 422 (HQ)24p from a Red camera.
Has anyone else experienced this green stripe issue and know a solution? I'd also appreciate any suggestions on how to improve the output quality when compressing in h.264.
Also, has anyone gone back to the previous version of Compressor included with FCP 6 (sorry, I don't know what version of Compressor was included w/ FCP 6)? If so, does it play nicely with FCP 7?

I may have just answered my own question with the green stripe artifact issue in Compressor 3.50. Thought I'd share it to see if it helps anyone else.
A true 16:9 frame means the height of the frame is 0.5625% of the length (9 / 16 = .5625). I used to compress material with a frame size of 500 X 282. Although this wasn't a true 16:9 frame, the difference didn't create noticeable distortion and previous versions of Compressor handled it ok. In 3.5 this deviation appears to create strange artifacts, in my experience, on the right edge of the screen in the form of a solid green stripe. When I moved my frame size to 512 x 288, the green stripe artifact went away. 288 / 512 = .5625, which is a true 16:9 frame.
I have no idea if this is just a coincidence or a solution. These are the results of what 30 min. of testing yielded. Perhaps it will be helpful to others in dealing w/ the strange line artifacts on the edges of frames.

Huge Network and Voice quality issues with Xperia T2 Ultra Dual

Same problem here after one month.
Voice cracking can't hear other party while having full signal strength.

Hi @Mayur9 Go to Call settings -> General -> and disable the "Slow talk option" if it is enabled. If the issue persists, boot your device in Safe mode to check what happens: http://talk.sonymobile.com/t5/FAQ/How-to-boot-your-phone-into-safe-mode/m-p/348008#U348008 Also I suggest you to first run a backup of the device information like this: http://talk.sonymobile.com/t5/FAQ/Backup-amp-Transfer/m-p/407253#U407253 Try performing a system repair using PC Companion: http://support.sonymobile.com/gb/tools/pc-companion/ This will factory reset your phone and erase all data. You might need tocharge your device for about 1-2 hours before so the device has at least 80% battery. Make sure the device is not connected to computer or laptop and turned off (sometimes you have to press power button and volume up until the device vibrates 3 times or on some devices you have to press the small button near the SIM tray or on the back of the device). Start PC Companion and select Support zone then Update my phone/tablet then in Blue Repair my phone/tablet and follow the on screen instructions - When prompted connect your phone still switched off holding volume down or back button - This should start the repair or reformat process. If you are using Windows 8/8.1 or a 64bit operating system then adjust the settings for PC Companion and run in compatibility mode and chose Windows 7 or XP. I hope can be useful. I'm at disposal for any question. Best Regards,

Voice Quality Issues on UC520

We have UC520 connected via SIP trunk to a provider. We have been experiencing bad quality when more than one line is connected to a conferencing system outside. So lets say more than one person call this conferencing system through the SIP trunk from multiple extensions. The audio when people are speaking is very choppy and robotic. As soon as all except one drop out, it becomes normal.
We have over 30 mbits/sec of bandwidth available at all time so I am not sure what could be causing this.
Have disabled VAD and looked at this article http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008019ab88.shtml
but cannot figure it out. ANy help would be appreciated.

You have poste in the wrong forum, anyay:
That indicates packed loss, even if you believe that you have the bandwidth, it can still happen. That is the naturn of the Internet, for which people still prefer POTS circuits for rock-solid service, or in some cases thye get dedicated data circuits for voice.
You can confirm if that is the case having multiple extensions. calling multiple external numbers and testing quality.
You can review UC500 configuration to optimize it, but it's unlikely that would solve the problem.
As a workaround for the conference problem, you can have a single extension call the conference service, and then have other phones conference to that call, not the external number directly.

Wireless voice quality issues with wlc 5508 7.0.98

Hi,
I am having random occurances of voice drops (one-way audio) during phone calls. WLC 5508 (7.0.98) , LAP1242AG (only G antenna present), and 7925G phones. coverage is excellent throughout the floor and its a confined office space. Its not happening always. I am seeing these logs , not sure if it is related. :
*apfReceiveTask: Feb 10 11:31:53.831: %RRM-3-RRM_LOGMSG: rrmChanUtils.c:290 RRM LOG: Airewave Director: Could not find valid channel lists for 802.11bg
*apfReceiveTask: Feb 10 11:31:33.356: %RRM-3-RRM_LOGMSG: rrmChanUtils.c:290 RRM LOG: Airewave Director: Could not find valid channel lists for 802.11bg
I have set DCA list to1,6 and 11. I tried disabling RRM and statically fixing the channels and power also. Still the issue is seen,
7925G firmware is 1.4.1
i tried to do linktest from the WLC to the phone, but link test is failed. linktest to a laptop works though. I have only mac filtering for the voice ssid.
any suggestions pls ?
regards
Joe

Hi Serge,
Thanks for the suggestions. I have taken care of all the settings. Problem is , the user is seated in his office cabin and using the 7925 and there is around 50db signal strength in his room from an AP which is just outside the cabin. And this doesnt occur often, when we go to check and make calls, everything is fine whereas the user says it happens sometimes in the morning, evening etc... randomly... where the fone goes blank while in a call.... since the problem never happens when we go to troubleshoot the issue, we really dont know what is going on. This is happening when calling PSTN.
AP's are not restarting and there is no logs indicating that. Wireless infrastructure looks very much OK cos there are other users who are not experiencing this problem and there is ample coverage all over the floor. I am baffled why only one user has this problem even when he is seated in his office cabin.
Tried changing phones also, but still remains.
Question : i tried to do a linktest from WLC to his phone, but it failed, while linktest to laptops are working fine. Does wlc linktest not working for wireless phones ?
regards
Joe

7925 Phones voice quality issues with wlc 5508 version 7.6

Hi all,
I have a mix environment with 1 WLC 5508 and more or less 6 sites with several Access Points ( all AIR-LAP1242AG and all in FlexConnect mode Hreap ) and several wireless phones (all CP7925G) . My Ap's have antennas 2.4GHz in all sites except 1 site ( the one i'm talking for now) with 2.4GHz and 5GHz , because of the problems we suggest to have all phone in A BAND (5Ghz) . In this site we force the phones just to A (802.11a only) , power safe NONE , Continuous scan mode , i fallow all in http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7925g/7_0/english/deployment/guide/7925dply.pdf , one thing i didn’t do it was apply QoS because I can’t apply several ios commands in that 2960 with lan lite ios.
So I have 3 SSID for data and 2 for VOICE ( one is common to all environment and supports 2.4 and 5Ghz , and other just to test one site , with only 5Ghz with “[WPA2][Auth(802.1X + CCKM)][Auth(FT 802.1X)]”, because I read that problem can be phone rooming , and in FlexConnect only have fast-roaming in a CCKM or a PKM key-management solution , but even like this the problem still exist) .
I have another sites with phones in 2.4GHz running well and others with problems.
i read here in forums lots and lots of people with similar problems , i tried several solutions ( WLC upgrades versions and phones firmware's , …. ) tried all my best of solving the problems mas now I’m running without any ideas , i did also some site surveys and detect some interferers that why I change phones to 5Ghz , and I run also WLC Config analyser to help me , but all without good success. Some changes solve temporarily problems , but after some time , people reclaim about problems.
I can open a TAC case but first I want be sure if did all already.
Can someone try to help me?
ip phone 7925G firmware : CP7925G-1.4.5SR1.3
CUCM 8.6.2
WLC    5508 : 7.6.120.0
AP : AIR-LAP1242AG-E-K9    7.6.120.0       (flexconnect)
Best Regards

Hi Serge,
Thanks for the suggestions. I have taken care of all the settings. Problem is , the user is seated in his office cabin and using the 7925 and there is around 50db signal strength in his room from an AP which is just outside the cabin. And this doesnt occur often, when we go to check and make calls, everything is fine whereas the user says it happens sometimes in the morning, evening etc... randomly... where the fone goes blank while in a call.... since the problem never happens when we go to troubleshoot the issue, we really dont know what is going on. This is happening when calling PSTN.
AP's are not restarting and there is no logs indicating that. Wireless infrastructure looks very much OK cos there are other users who are not experiencing this problem and there is ample coverage all over the floor. I am baffled why only one user has this problem even when he is seated in his office cabin.
Tried changing phones also, but still remains.
Question : i tried to do a linktest from WLC to his phone, but it failed, while linktest to laptops are working fine. Does wlc linktest not working for wireless phones ?
regards
Joe

Itunes Video Quality issues and crash in Windows vista

When using itunes listening to music works great, however when i play music videos purchased from the store they keep on skiping and after a while itunes will close and a message from windows will appear saying the software stopped working. when the video is playing the sound is fine and plays quite happily in the background however the video freezes and jumps.

Have you tried this article:
Troubleshooting iTunes for Windows Vista video playback performance issues
http://docs.info.apple.com/article.html?artnum=305392

Fragmenting packets over Ethernet to improve voice quality

Setup:
1750--Ethernet---Satellite Receiver---128K Sat Link---Sat. Recvr--Ether---1750
Question:
Since the Satellite Link is only 128K there will be a problem of Serialization Delay. Is there a way to fragment packets and Interleave over ethernet ? The same way it is done over Frame Relay or MLPPP.
In this setup how does one lower down "Serialization Delay" ?

rocampo,
With a Satellite link, most the latency will be travel time and not serialization delay. I would expect VoIP quality to be an issue, since the users will have to tolerate long delays and have to wait to make sure the person on the other end has actually finished speaking before starting to speak. VoIP packets are generally small enough that they are already at or near Ethernet minimum packet size of 64 KBytes, so they are not fragmented on Ethernet. However, you may want to look into the QOS / COS capabilities of your LAN switches. But the real issue for you will be the large amount of latency across the satellite link. Be sure to use QOS on the satellite link, to send along VoIP packets on a preferred basis over other less time sensitive traffic. And you may want to see if you can use RTP header compression on the Satellite link to shrink the size of VoIP frames.
Regards,
Rob Bristow
AT&T Solutions
CCIE #3335

Lync to PSTN quality Issues, no issues to be found in reports

We have a lync 2013 install recently completed however we are experiencing voice quality issues. It appears to be specifically calls between Lync clients and PSTN and additionally the issues are only heard on the PSTN side (Lync side sounds great). This
is regardless if it was a PSTN initiated or Lync initiated call.Reports of quality issues are given as "sounding watery", "Robotic", "Fading in and out", and "Momentary loss of sound". It it VERY consistent.
Our Lync to Lync calls are perfect (internally and federated), or if its not good quality it is easily diagnosed.
We are running 2 lync edge servers, two front end. currently only have 40 users. We have a 40mb internet circuit. QOS is enabled on the edge router\core switch and we have had the QOS investigated and does not appear to be the issue (I read another forum
post indicating QOS causing issues and we contracted a firm to verify QOS). We are using Intelepeer as a SIP trunk and it terminates to an Audiocodes Mediant 1000. The Mediant 1000 connects directly to the internet.
When we look at the call statistics there are no issues to be found. We have opened tickets with Intelepeer and they do not show any issues either.
I am starting to feel a bit lost as to where to look for help. Please let me know what additional info you need and I will be grateful and happy to give more info.

Assuming you are not encrypting traffic between Mediation and gateway, you can use Wireshark to capture the traffic of the “internal” leg. Later you can “decode” and listen the actual call which will be the flow Lync <-> Mediation <-> Gateway.
If the quality is good, mirror switchport and Wireshark the traffic between Gateway and IntelePeer. This way you can identify the leg that introduces the issue. Just a place to start...
Drago
http://www.lynclog.com

Unity Connection Voice Mail Issue

Hi,
I have a weird Unity Connection Voice Mail issue and would appriciate any help i can get. Many thanks in advance.
I have Unity connection 7.0 and CUCM 7.0 integrated in a lab enviroment and here is what happens.
When I place a call internally say from 2001 to 2002, things work as expected, 2002 rings and it goes to voicemail where I can leave voicemail and listen to it from 2002.
However, if I place call from the PSTN to the same number 2002 (or any other number in other sites etc) the call again rings in 2002 and goes to voicemail, unity cnx plays the greetings for 2002 and says record your message as usual. Everything up to point is fine, then when the time on the PSTN phone is showing around 14 seconds into the call unity starts playing, "to send this message press one", if I press one nothing happens.
I have rebuilt unity and cucm, even just configured the bare minimum in the lab and still getting the same result. I tried calling from E1 connection and T1 connection but with the same results.
I have run out of ideas...
===============================
Here is a call trace from an internal call:
CallData, 1, CallerId=2003, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=8B7859FD6C16417A9A07F507418DD25B, CallerName=, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 2003, AttemptForward
State, 1, 2003, State - AttemptForward.cde!Dummy
State, 1, 2003, Event is [NULL]
Application, 1, 2003, PHTransfer
State, 1, 2003, State - PHTransfer.cde!LoadInfo
State, 1, 2003, Event is [TrueEvent]
Application, 1, 2003, PHGreeting
State, 1, 2003, State - PHGreeting.cde!PlayGreeting
Display, 1, 2003, Call answered if needed
Display, 1, 2003, Playing greeting for Subscriber: hq2
Display, 1, 2003, No DTMF received
Display, 1, 2003, Playing greeting for Subscriber: hq2
State, 1, 2003, Event is [RecordMsgEvent]
State, 1, 2003, State - PHGreeting.cde!RecordMsg
State, 1, 2003, Event is [NULL]
State, 1, 2003, State - PHGreeting.cde!RunEditMsg
Application, 1, 2003, -->MessageEditing
State, 1, 2003, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 2003, Event is [EditMessageMenuEvent]
State, 1, 2003, State - MessageEditing.cde!PlayEditMenu
State, 1, 2003, Event is [HangupEvent]
State, 1, 2003, State - MessageEditing.cde!CheckMsgLength
State, 1, 2003, Event is [ManyEvent]
State, 1, 2003, State - MessageEditing.cde!SendMsg
State, 1, 2003, Event is [TrueEvent]
State, 1, 2003, State - MessageEditing.cde!ConfirmSend
State, 1, 2003, Event is [HangupEvent]
Application, 1, 2003, <--MessageEditing
State, 1, 2003, Event is [HangupEvent]
Display, 1, 2003, Idle
Display, 1, , Dialing (MWI) '2002'
Display, 1, , Idle
and here is a trace from an external (PSTN) call
Trying 142.100.64.13, 5000 ... Open
CallData, 1, CallerId=911, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=CA3DFD90846C4FE7B0D68298A7698287, CallerName=PSTN Emergency, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 911, AttemptForward
State, 1, 911, State - AttemptForward.cde!Dummy
State, 1, 911, Event is [NULL]
Application, 1, 911, PHTransfer
State, 1, 911, State - PHTransfer.cde!LoadInfo
State, 1, 911, Event is [TrueEvent]
Application, 1, 911, PHGreeting
State, 1, 911, State - PHGreeting.cde!PlayGreeting
Display, 1, 911, Call answered if needed
Display, 1, 911, Playing greeting for Subscriber: hq2
Display, 1, 911, No DTMF received
Display, 1, 911, Playing greeting for Subscriber: hq2
State, 1, 911, Event is [RecordMsgEvent]
State, 1, 911, State - PHGreeting.cde!RecordMsg
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!RunEditMsg
Application, 1, 911, -->MessageEditing
State, 1, 911, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 911, Event is [EditMessageMenuEvent]
State, 1, 911, State - MessageEditing.cde!PlayEditMenu
State, 1, 911, Event is [HangupEvent]
State, 1, 911, State - MessageEditing.cde!CheckMsgLength
State, 1, 911, Event is [NULL]
Application, 1, 911, <--MessageEditing
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!AfterMsg
State, 1, 911, Event is [NULL]
Display, 1, 911, Idle

Sounds like one way audio from PSTN to your Unity Connection, couple of things to check:
1. ensure your protocols are bound properly on the GW, i.e. SIP/H323/MGCP
2. Make sure IP routing is OK between Unity and the voice gateway
HTH,
Chris

Print quality issues

Hello
We are having print quality issue when printing PDF's to our Canon iR C2880/C3380 UFR II. This was the most recent print driver from canons website which was download about 6 weeks ago. version 2.9. At one time we were having other pring quality issues and we changed from the post script driver to the ufr ii which solved the problem at the time.
This past saturday we upgrade the operating systems on the systems in question and they are now running windows 7 64 bit with service pack 1 installed. image 1 is the orignal pdf that was printed with a 3rd party pdf printer(PDF Xchange view version 2.5) image 2 is the same document that was printed using acrobat reader 10.1.2
the options for the print were pretty much identical between the program. Print as image was not selected. 1 thing to note with the acrobat when print as image was selected there was a slight improvement in the print quality.
Any help would greatly be appreciated.
image 1
image2

Hi, nd a warm welcome to the forums!
Off the bat, it looks like maybe bad RAM/ROM in the Printer, or corrupt Font.
Can you have another Computer jook to it and print for a test?
Can you try Printing from Adobe Reader...
http://www.adobe.com/products/acrobat/readstep2.html
Can you in Preview, Save As a picture format for a test?

IPhone 4s - Battery issue and poor voice quality

I had bought my iPhone in May this year. It was working fine till i think a week back battery started to drain quickly at 10% every hour. I have configured office exchange account for mail and accessing global exchange contacts, Google account as exchange for contact sync and iCloud. i dont keep any contacts on exchange, keep in Gmail only. Was Android user earlier.
I read a lot of posts and tried all solutions such as turning off timezone service in location, turn off and delete all email accounts, kept manual sync for emails, ran the battery down till switch off and charging back to 100%, removed apps i doubted could be syncing data in background, turned off locaiton service totally, turned off calendar, turned calendar notifications, turned of iCloud reminder, notes but no effect. Wifi and Siri are also off.
I notice today that I am getting small circle near operator name rather than E symbol (for edge)and at the same time voice quality of person on other end is bad too, can hear small cracking , clicking sounds, voice breaks. Otther person can hear me properly. I get back E symbol in some different place than my desk. I havent changed my desk so thats not a possibility either.
Someone had suggested queued network thread with exchange can cause the issue so Installed System Activity Monitor app but it doesn't show battery graph or which process is taking CPU or network time. Moreover, it shows 4-5% less battery level than on top in disharging mode !!
Used to get good stats of what is eating the battery in my old HTC Desire but iPhone doesn't offer such feature to diagnosis. Ideally should not be required if phone works as expected.
What do I need to do to get battery life and voice quality back? I will try voice again at home just to rule out bad network at work.
Thanks,
Ashish

Well, most of what you did is black magic, and has nothing to do with battery drain. But we have no way of knowing how your battery life is below what you think it should be. What are you using it for, what apps do you have, and how much Usage life do you get (settings/general/usage).

IPad 2 Video playback Quality Issue (Low Contrast and grayish black)

Does anyone notice that the video playback quality is really bad on the iPad 2? If you try to play any video (youtube, apple trailer, your own converted video) on the iPad 2, any black images will turn grayish... Just put your iPhone, iPod touch, or even the older iPad next to it, then you will notice a BIG difference. The color is similar to what happened when you set the gamma incorrectly on a display.
Also, you will notice this more if you change the orientation of the device, the contrast will be corrected for a fraction of a second.
I've tried this on other iPad 2 and they all share the same problem. I tried to talk to the Apple Genius Bar and they refused to give me a proper answer on this, instead simply claiming that is just the way it is... I wanted to refund this but I've passed the 14 days rule and also I got it from a authorized retailer and so Apple just won't take it back.
I believe it can be fixed by software... Any idea? or I shall throw it away T_T

I'm on the phone with apple support right now. I work at a post production house, and we/our clients keep having this issue - everything looks great on screen and on the ipad 1, and theres a huge gamma shift on the ipad 2. they need to fix this ASAP. like Jls21 already said, a lot of us use ipads for work purposes - unacceptable!
this thread ( https://discussions.apple.com/message/15100980#15100980 ) seems to blame the h.264 codec/compression settings…? does anyone know of another codec that prevents this gamma shift on the ipad 2?
this is from the tech specs page:
Video formats supported: H.264 video up to 720p, 30 frames per second, Main Profile level 3.1 with AAC-LC audio up to 160 Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats; MPEG-4 video, up to 2.5 Mbps, 640 by 480 pixels, 30 frames per second, Simple Profile with AAC-LC audio up to 160 Kbps per channel, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats; Motion JPEG (M-JPEG) up to 35 Mbps, 1280 by 720 pixels, 30 frames per second, audio in ulaw, PCM stereo audio in .avi file format
is there anything that jumps out to anyone regarding workarounds for this issue?
we've used both Compressor and MPEG Streamclip - there's no quality difference between them, in case anyone is trying that.
okay so, i was transferred to a senior technical rep and he looked it up - apparently apple is getting a bunch of complaints about this issue and they've submitted it to their developers. the more people that call and complain, the higher up on the priority list it goes to be fixed in a software update. there is no workaround known, and it's a standard issue across all ipad 2's.

Quality, Export, and Compression help (pic included)

I'm working on an edit of my RTW trip and am struggling to figure out some quality issues. Video was captured by 3 cameras (a JVC everio, a GoPro Hero 2, and a Droid 4). All record in HD (720 or 1080). I figured I would use the lowest common denominator and converted all the video in MPEG streamclip to be 1280x720 HD, using Apple Intermediate Codec and 100% quality. I have the whole timeline basically finished in FCE, but I'm trying to refine it now and I'm really bothered by the quality. It doesn't look HD at all...it looks really pixelated. Below is a screenshot of my viewer, which is at 100%. Note the edges where the blue arrows point-they're all jagged.
It looks decent at 100% but that is not even 1/6th of my screen. If I go up to 200% it looks like crap. Is this just because I'm editing on a 26" iMac and my screen is so big that I need 1080p or 4K to look good at full screen? When I cropped the 100% screenshot in Preview I noticed that it is 640x360 pixels...exactly half of 1280x720 like I would expect. Why is that? How do I get the quality to be like it should? Am I losing it in MPEG Streamclip or Final Cut? I haven't even started the export battle, which always infuriates me with how bad the quality comes out if the video is a reasonable file size (This video is 5m40s, I feel like 100MB or less is reasonable for streaming right?)
Thanks a ton for the help. I really want to finish this edit this week but want to make sure I do my trip of a lifetime justice.

1.) Any advice for resizing the 1080 clips?
Go to the first clip in your timeline that is 1080 and looks incorrectly sized. Double click on that clip in the timeline, you will see the clip open in the Viewer. Once it does that, in the Viewer, click on the Motion tab, and then adjust the Scale paddle until the frame is just filled. Once that is done, type Command + C to copy the clip and the settings adjustment you made.
Now go to the next instance of a 1080 clip in your timeline and click on it to select it. If there a bunch together you can hold down the shift key while you click on the clips and select as many as you want at a time. Once you have made the selection, type Option + V to paste the attributes from the clip you had just copied. The Paste Attributes window should open, and you want to check the Basic Motion box and keep everything else unchecked. Then click OK, and the clips you had selected should now all be resized.
2.) Some of the clips still have blurry frames. Now that I have the sequence totally edited and don't need to trim/move clips all the time, if I use the raw source video (not optimized for Final Cut via MPEG streamclip and the Apple Intermediate Codec) in Final Cut, will the quality be better or will it be the same/not work at all?
Final Cut Express will only work with HD material in Apple Intermediate codec. You were smart to convert the footage prior to editing. Are the clips that look blurry the 1080 clips? At any rate, you may need to render your timeline, see if any colored render bars are showing on the timeline (rendering will occur automatically when you export your finished movie).
MtD

Voice Quality Issues and RTP Header Compression

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