On 27/11/11 18:10, Phil Quesinberry wrote:
A friend of mine dialed into one of my conference rooms with only the GSM codec enabled on his mobile SIP client (Linphone). Despite the fact that I offered GSM as one of the codecs, FS only appeared to offer G711u and the call failed. When he set G711u as an option on his end, he was able to connect. Late negotiation is enabled.
The SDP from the mobile SIP client is bogus, it offers codec 0 (PCMU) but tries to redefine it as GSM. FreeSWITCH accepts the offer of PCMU and negotiates it - but then the callers client sends/expects invalid media: m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/22050 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 http://www.iana.org/assignments/rtp-parameters 0 PCMU/8000 3 GSM/8000 It should be using 96-126 for GSM at other bitrates... Regards,