Wireshark-dev: Re: [Wireshark-dev] no sound during playback of amr call
From: wireshark-dev-bounces@xxxxxxxxxxxxx [mailto:wireshark-dev-bounces@xxxxxxxxxxxxx]
On Behalf Of Rahul Rohit >Hi, >What I understood is that to check playback feature for every standard payload type of RTP we need codecs specific to that standard .By default wireshark has codec only for
RTP payload type 0 and >8(G.711 A class and U class) and doesn’t have codecs for remaining payload types
> (including dynamic payload stream 96). Payload type 0 – 95 have fixed codec mapping, 96 – 127 are dynamic – e.g the payload type to codec mapping is dynamic and decided by out-of-band means, in SIP by SDP in SIP signals. So payload type 96 MAY be G711 depending what’s been signaled (or set) between the two RTP end points but most probably not as G711 have a fixed payload type. So you can’t say that Wireshark does not have a codec for PT 96, it does not have a codec for the media sent as payload type 96 in that particular call. >Is this true that we can’t check the playback functionality in Wireshark without having a licensed/patented codec for any particular standard of RTP other than G.711 A class
and U class ? To play RTP content you need a codec, and the only available codecs in Wireshark are G711a and G711u, so yes.
>P.S.: In Wireshark we’ve added a preference setting in GUI to receive RTP packets over UDP depending on the port values. We are not using “decode >as” functionality. Thanks Rahul Rohit From:
wireshark-dev-bounces@xxxxxxxxxxxxx [mailto:wireshark-dev-bounces@xxxxxxxxxxxxx]
On Behalf Of Anders Broman Hi, I’m not sure what you are after. To playback an (audio) RTP stream the player must support the codec used. Currently Wireshark does only support playback of G711A/U (see
http://wiki.wireshark.org/RTP_statistics and
http://wiki.wireshark.org/VoIP_calls ) you can add codecs to C:\wireshark\plugins\easy_codec or C:\wireshark\codecs Unfortunately I don’t know the details of doing it. Note that many codecs come with licenses an patents that make it difficult to use them in OpenSource. From “RTP dynamic payload stream(PT_UNDF_96)” I guess that you don’t have the setup information of the RTP stream in your trace and have done decode as RTP. As far as I know currently there is no way to “tell”
Wireshark which codec is used for payload type > 96. Perhaps something like that should be added. An UAT table to specify src/dst IP/port and PT = xxx in the RTP dissector perhaps. Note that in theory Payload type mapping might differ per call. Device A might propose AMR as PT = 96 and device B might use 101. Some payload dissectors allow you to specify which PT they are tied to, check the preferences of the AMR dissector. Regards Anders From:
wireshark-dev-bounces@xxxxxxxxxxxxx [mailto:wireshark-dev-bounces@xxxxxxxxxxxxx]
On Behalf Of Rahul Rohit Hi, Is there any way by which Wireshark be made to support play back for the RTP dynamic payload stream(PT_UNDF_96)by performing any change in the code or Wireshark. Regards Rahul Rohit From:
wireshark-dev-bounces@xxxxxxxxxxxxx [mailto:wireshark-dev-bounces@xxxxxxxxxxxxx]
On Behalf Of Rahul Rohit Hi, We are playing back an AMR voice call in Wireshark using telephony option but unfortunately no sound is audible. Could somebody please help us as to what might be the probable reason for this. Regards Rahul Rohit "DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential
information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are
strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus."
"DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential
information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are
strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus."
"DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential
information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are
strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus."
|
- References:
- [Wireshark-dev] no sound during playback of amr call
- From: Rahul Rohit
- Re: [Wireshark-dev] no sound during playback of amr call
- From: Rahul Rohit
- Re: [Wireshark-dev] no sound during playback of amr call
- From: Anders Broman
- Re: [Wireshark-dev] no sound during playback of amr call
- From: Rahul Rohit
- [Wireshark-dev] no sound during playback of amr call
- Prev by Date: Re: [Wireshark-dev] no sound during playback of amr call
- Next by Date: [Wireshark-dev] controlling the display of the leading bit string when bitmask != 0
- Previous by thread: Re: [Wireshark-dev] no sound during playback of amr call
- Next by thread: [Wireshark-dev] Lua: Error during loading:
- Index(es):