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  1. #1
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    APS recorder and player settings

    Hi,

    I am developing a VoIP client for E61 (APS version 2.40). I am able to record and play voice during VoIP call with the help of STREAMER sample. I am facing RTP Jitter problem at remote Rx end during call.

    I tried the different combinations of
    TMdaPriorityPreference(0x05210001),
    TMdaPriorityPreference(0x05220001),
    EMdaPriorityPreferenceNone,
    EMdaPriorityPreferenceTime,
    EMdaPriorityPreferenceQuality and
    EMdaPriorityPreferenceTimeAndQuality for RSetting and PSetting preferrence.
    But in none of the case i could solve the RTP Jitter issue at remote end. In some cases remote end can hear the voice clearly but no voice could be heard at local side.

    Can anybody tell me at what preferrence should i keep the recorder and player to solve the jitter issue? Is there any other parameters that can be checked to smoothen the voice streaming?

    Thanks
    Rajat

  2. #2
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    Re: APS recorder and player settings

    hi,


    [I am able to record and play voice during VoIP call with the help of STREAMER sample. I am facing RTP Jitter problem at remote Rx end during call.]
    does it mean you have used STREAMER sample for VoIP call?
    well, i am not very knowledgeable about RTP jitter.
    how are you experiencing RTP jitter problem?
    did you observer actual packet delay through ethereal capture?
    did you observe packet delay due to problem with voice playing/recording?

    [But in none of the case i could solve the RTP Jitter issue at remote end. In some cases remote end can hear the voice clearly but no voice could be heard at local side.]
    is voice not heard at all at local side?

    (For G.711/iLBC codecs.)
    for NO VOICE at local side: ===> check whether you have added 2-byte header (required for the APSServer for playing the packet) to the received RTP data before sending it to the PlayQueue (refer Section 10. Appendix of APS design document)

    similarly, when sending the sampled data to the remote end as RTP, check whether the first 2-bytes is removed from the data received by the Recorder.

    hope this helps.
    rgds,
    vasu

  3. #3
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    Re: APS recorder and player settings

    Hi Vasu,

    Thanks for the reply.

    The answers of your question is as follows.

    does it mean you have used STREAMER sample for VoIP call?
    Yes.

    how are you experiencing RTP jitter problem?
    Using the WireShark

    did you observer actual packet delay through ethereal capture?
    Yes

    did you observe packet delay due to problem with voice playing/recording?
    I found that APS keeps the mean of voice data recording at 20ms. But there is some problem in voice data parsing and sending, causing jitter.

    is voice not heard at all at local side?
    Yes. But this is due to the preferrence changed for RSettings and PSettings. I found in one post that the preference should be used provided in streamer sample.

    for NO VOICE at local side: ===> check whether you have added 2-byte header
    similarly, when sending the sampled data to the remote end as RTP, check whether the first 2-bytes is removed from the data received by the Recorder.
    Checked

    Currently i am worknig on replacing the RTP TX by polling instead of callback. Are there any other options to be checked to resolve RTP Jitter?

    Best Regards
    Rajat

  4. #4
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    Re: APS recorder and player settings

    hi,

    As of now, i have not experienced jitter problem with the below mentioned settings.
    It could be because
    - i have tested in a LAN environment(intranet).
    - as well my testing being very minimal.

    which codec you are using?
    what is the timestamp interval of sent/received RTP packets?
    how many bytes(UDP) are being sent/received for each RTP packet?

    if there is no problem with sent/received RTP packets and it is genuinely a jitter problem, then 3rd party applications are supposed to handle the Jitter Management.

    http://www.forum.nokia.com/info/sw.n..._0_en.pdf.html

    rgds,
    vasu

  5. #5
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    Re: APS recorder and player settings

    Hi,

    which codec you are using?
    G711

    what is the timestamp interval of sent/received RTP packets?
    It varies. I am using NTickCount as timestamp.

    how many bytes(UDP) are being sent/received for each RTP packet
    114 Bytes sent/134 Bytes received


    Best Regards
    Rajat

  6. #6
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    Re: APS recorder and player settings

    hi,

    what is the timestamp interval of sent/received RTP packets?
    It varies. I am using NTickCount as timestamp.

    I am asking about difference of timestamp value between two consecutive RTP packets you send/receive.
    OR what is the packetization time? i.e., G711 packet is sampled for how many milliseconds?

    how many bytes(UDP) are being sent/received for each RTP packet
    114 Bytes sent/134 Bytes received
    114 - 20(udp-8 + rtp-12) = 94 bytes (G711 data)
    134 - 20(udp-8 + rtp-12) = 114 bytes (G711 data)
    I doubt there is some problem with the bytes sent and received.
    Probably, the Timestamp interval would give an indication of your problem.

    If data is sampled at 10 ms using G711, it uses 80 bytes.
    Total packet size would be 80 + 20 = 100 bytes (UDP packet)
    For 20 ms, G711 packet size would be 180 bytes (UDP packet)
    For 30 ms, G711 packet size would be 240 bytes (UDP packet)

    rgds,
    vasu

    P.S.:-> i would really appreciate your comments on the following.
    are you using Symbian SIP Stack in S60?
    if yes, whether one needs to apply for API partnering request to get incoming VoIP call notification on E61?(which i am not getting if m-line contains "audio") OR am I missing anything?
    if no, have you ported SIP stack?

  7. #7
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    Re: APS recorder and player settings

    Hi,

    The timestamp interval is 160, as G711 packets are sampled for 20ms. And the voice data sent/received is of 160 bytes.

    Finally i got the solution of TX Jitter issue. I changed the packet sending mechanism to timer based instead of signalling based.

    _____________________________________________________________
    *Are you using Symbian SIP Stack in S60?
    Yes
    *Whether one needs to apply for API partnering request to get incoming VoIP call notification on E61?
    No
    *If no, have you ported SIP stack?
    I am using Nokia SIP Stack.
    ------------------------------------------------------------

    Best Regards
    Rajat
    Last edited by raj_rr7; 2008-01-10 at 05:49. Reason: RX kept instead of TX

  8. #8
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    Re: APS recorder and player settings

    Hi,

    The timestamp interval is 160, as G711 packets are sampled for 20ms. And the voice data sent/received is of 160 bytes.

    Finally i got the solution of TX Jitter issue. I changed the packet sending mechanism to timer based instead of signalling based.

    _____________________________________________________________
    *Are you using Symbian SIP Stack in S60?
    I am using the default SIP Stack comes with Nokia phones.
    *Whether one needs to apply for API partnering request to get incoming VoIP call notification on E61?
    No
    *If no, have you ported SIP stack?
    No
    ------------------------------------------------------------

    Best Regards
    Rajat
    Last edited by raj_rr7; 2008-01-10 at 05:50. Reason: RX kept instead of TX

  9. #9
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    Re: APS recorder and player settings

    Quote Originally Posted by raj_rr7 View Post
    Hi,

    Finally i got the solution of RX Jitter issue. I changed the packet sending mechanism to timer based instead of signalling based.

    Best Regards
    Rajat
    Just out of curiosity, Can you elaborate more on this? What is polling timer interval you are managing?

    Are the end devices are always Nokia?

    Have you turned on CNG/VAD?



    Thanks
    santhosh.

  10. #10
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    Re: APS recorder and player settings

    Hi Santosh,

    The jitter issue is appearing on the remote end during a VoIP call. If the remote end is NOKIA phone then it handles the jitter gracefully. I was facing the jitter issue when calling from E61 to Windows Mobile phone. And it happened due to Transmission/TX delay at E61 varies.

    Currently the time interval is 15ms. I tried with 20ms but found some issue in User::After. It sends the packets at 30ms of delay.

    I have not played with CNG/VAD parameters yet.

    Regards
    Rajat

  11. #11
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    Re: APS recorder and player settings

    Hi!

    What is TX jitter? How to implement it with streamer example given by nokia. Please let me know what is polling mechanism? How to implement it?

    -Anil

  12. #12
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    Re: APS recorder and player settings

    Hi Anil,

    TX Jitter is time variations that happen due to irregular packet transmission.
    With nokia sample, a separate queue needs to be created that can hold the APS data before transmitting. This data can be sent at regular interval.
    Polling mechanism is to poll at regular interval. In case of separate queue application can fetch the data from queue at a particular interval(10 ms/20 ms).

    Regards
    Rajat

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