[Freeswitch-users] static rtpmap entries are not shown in outbound invites

Tue Mar 8 16:07:44 MSK 2011

Hi Steven,

Ah yes, I knew it meant mono. But I found the problem..

* The call that worked properly with G729/8000 was on a sip profile that had G729 allowed in inbound-codec-prefs
* The call that didn't work was on a second profile that apparently didn't have G729 set for inbound-codec-prefs (I did specify it in the xml, but only did a reload on the profile which isn't sufficient - I should've restarted it -- doh)

Thanks all & kind regards,

Leon

On Mar 7, 2011, at 6:57 PM, Steven Ayre wrote:

>> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):
> 
> That means 1 channel (mono), which is the default. Although optional
> it is correct and *should* be handled fine...
> 
> -Steve
> 
> 
> 
> On 7 March 2011 13:59, Leon de Rooij <leon at scarlet-internet.nl> wrote:
>> Hi Brian,
>> 
>> Thanks, that works :-)
>> 
>> I got one last question about sdp, would you know why chan var ep_codec_string isn't set after an incoming invite with the following sdp:
>> 
>> v=0.
>> o=FOOBAR 123456 654321 IN IP4 1.2.3.4.
>> s=-.
>> c=IN IP4 1.2.3.4.
>> t=0 0.
>> m=audio 22262 RTP/AVP 18 101.
>> a=rtpmap:18 G729/8000/1.
>> a=rtpmap:101 telephone-event/8000/1.
>> a=fmtp:101 0-15.
>> 
>> Can it be the trailing /1 (encoding parameters) on both rtpmaps ? - According to parse_rtpmap() in libsofia it should be parsed alright, but still, ep_codec_string gets filled when I receive an SDP like this (from another endpoint):
>> 
>> v=0.
>> o=BARFOO 20494030 20494030 IN IP4 4.3.2.1.
>> s=-.
>> c=IN IP4 4.3.2.1.
>> t=0 0.
>> m=audio 50000 RTP/AVP 18 100.
>> a=rtpmap:18 G729/8000.
>> a=sendrecv.
>> a=ptime:20.
>> a=rtpmap:100 telephone-event/8000.
>> a=fmtp:100 0-15.
>> 
>> 
>> (btw, I'm having inbound- and outbound-codec-prefs on the incoming profile set as: "G7221 at 32000h,G7221 at 16000h,G722,PCMA,PCMU,G729,GSM" and am doing inbound-late-negotiation="true")
>> 
>> 
>> Thanks,
>> 
>> Leon
>> 
>> 
>> 
>> 
>> On Mar 4, 2011, at 7:09 PM, Brian West wrote:
>> 
>>> verbose_sdp=true
>>> 
>>> docs/ChangeLog: mod_sofia: apparently some sip device vendors did not read the RFC (who knew?) adding verbose_sdp=true var to add needless a= lines for standard iana codecs that explicitly do not require them (r:6c4f49a8)
>>> 
>>> /b
>>> 
>>> On Mar 4, 2011, at 12:06 PM, Leon de Rooij wrote:
>>> 
>>>> Can someone help me with this ?
>>> 
>>> 
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>> 
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