[Freeswitch-users] SIP over Websocket VS SIP over TCP

[Adam Ben-Ayoun]

Update: We integrated pjsip with WebRTC on iOS, we are using pjsip for the
signalling part and passing the SDP back and forth to and from WebRTC.

Our current issue is that after initializing the call we can't hear any
audio (should hear moh at this point), everything looks OK on Freeswitch
until we start seeing alot of those errors:
2015-01-16 20:25:31.111260 [ERR] mod_opus.c:418 Decoder Error: corrupted
stream fs:4080 plc:0!
2015-01-16 20:25:31.111260 [ERR] switch_core_io.c:697 Codec RAW Signed
Linear (16 bit) decoder error! [9]

Any idea what might be wrong here?

Thanks,
Adam


On 15 January 2015 at 15:27, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
wrote:

> In our case, all clients are calling to the conference on FreeSwitch,
> hence, no registration is required at the moment, but it definitely opens
> up other options for future.
>
> On 10 January 2015 at 20:38, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
>
>> Great to hear that. Thanks again.
>>
>> On 10 January 2015 at 20:35, Carlos Ruiz Díaz <carlos.ruizdiaz at gmail.com>
>> wrote:
>>
>>> WebRTC doesn't specify a signalling protocol. This means that you can
>>> use SIP over any transport you want to carry the webRTC enabled SDP.
>>>
>>> FS will receive the SDP, detect that has a RTP/SAVPF profile and start
>>> handling it accordingly.
>>>
>>> Take for example Jitsi or IMSDroid, they both support webRTC and do SIP
>>> over UDP/TCP/TLS.
>>>
>>> Regards,
>>> Carlos
>>> On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>>
>>>> Thanks Anthony. I assume that means I can use SIP over TCP/TLS for
>>>> signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work
>>>> when communicating with FS (when stuff like fingerprint, etc)?
>>>>
>>>> On 10 January 2015 at 19:49, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> The WebRTC media engine is driven completely by the SDP, the transport
>>>>> will not make any difference.
>>>>>
>>>>>
>>>>> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun <
>>>>> adam.ben.ayoun1 at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> We are developing a mobile client that will use the WebRTC media
>>>>>> stack and Freeswitch as an MCU (only for conference calls). My question is,
>>>>>> since we build a native app, can we use SIP over TCP for signalling? In
>>>>>> other words, if Freeswitch receives the WebRTC kind of SDP, will it be able
>>>>>> to communicate in the same way as if we were using the SIP over Websocket
>>>>>> (the other Freeswitch option)? Any corner cases/considerations with this?
>>>>>> Our goal is to avoid implementing SIP over Websocket on the client as much
>>>>>> as possible.
>>>>>>
>>>>>> Thanks,
>>>>>> Adam
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
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>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>>>>
>>>>> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
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>>>>>
>>>>>
>>>>>
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>>>>
>>>>
>>>>
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>>>
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>>
>>
>
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